Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 7604                                      Ericsson
Category: Informational                                          T. Zeng
ISSN: 2070-1721                                         PacketVideo Corp
                                                          September 2015
Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 7604                                      Ericsson
Category: Informational                                          T. Zeng
ISSN: 2070-1721                                         PacketVideo Corp
                                                          September 2015

Comparison of Different NAT Traversal Techniques for Media Controlled by the Real-Time Streaming Protocol (RTSP)




This document describes several Network Address Translator (NAT) traversal techniques that were considered to be used for establishing the RTP media flows controlled by the Real-Time Streaming Protocol (RTSP). Each technique includes a description of how it would be used, the security implications of using it, and any other deployment considerations it has. There are also discussions on how NAT traversal techniques relate to firewalls and how each technique can be applied in different use cases. These findings were used when selecting the NAT traversal for RTSP 2.0, which is specified in a separate document.

本文档描述了几种网络地址转换器(NAT)遍历技术,这些技术被认为用于建立由实时流协议(RTSP)控制的RTP媒体流。每种技术都包括如何使用它的描述、使用它的安全含义以及它所具有的任何其他部署注意事项。还讨论了NAT穿越技术如何与防火墙相关,以及每种技术如何应用于不同的用例。在为RTSP 2.0选择NAT遍历时使用了这些发现,这在单独的文档中有详细说明。

Status of This Memo


This document is not an Internet Standards Track specification; it is published for informational purposes.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 5741.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。并非IESG批准的所有文件都适用于任何级别的互联网标准;见RFC 5741第2节。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2015 IETF信托基金和确定为文件作者的人员。版权所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

本文件受BCP 78和IETF信托有关IETF文件的法律规定的约束(自本文件出版之日起生效。请仔细阅读这些文件,因为它们描述了您对本文件的权利和限制。从本文件中提取的代码组件必须包括信托法律条款第4.e节中所述的简化BSD许可证文本,并提供简化BSD许可证中所述的无担保。

Table of Contents


   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Network Address Translators . . . . . . . . . . . . . . .   5
     1.2.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . .   6
     1.3.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   7
   2.  Detecting the Loss of NAT Mappings  . . . . . . . . . . . . .   8
   3.  Requirements on Solutions . . . . . . . . . . . . . . . . . .   9
   4.  NAT-Traversal Techniques  . . . . . . . . . . . . . . . . . .  10
     4.1.  Stand-Alone STUN  . . . . . . . . . . . . . . . . . . . .  11
       4.1.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  11
       4.1.2.  Using STUN to Traverse NAT without Server
               Modifications . . . . . . . . . . . . . . . . . . . .  11
       4.1.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  14
       4.1.4.  Deployment Considerations . . . . . . . . . . . . . .  14
       4.1.5.  Security Considerations . . . . . . . . . . . . . . .  15
     4.2.  Server Embedded STUN  . . . . . . . . . . . . . . . . . .  16
       4.2.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  16
       4.2.2.  Embedding STUN in RTSP  . . . . . . . . . . . . . . .  16
       4.2.3.  Discussion on Co-located STUN Server  . . . . . . . .  17
       4.2.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  17
       4.2.5.  Deployment Considerations . . . . . . . . . . . . . .  18
       4.2.6.  Security Considerations . . . . . . . . . . . . . . .  19
     4.3.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  19
       4.3.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  19
       4.3.2.  Using ICE in RTSP . . . . . . . . . . . . . . . . . .  20
       4.3.3.  Implementation Burden of ICE  . . . . . . . . . . . .  21
       4.3.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  22
       4.3.5.  Deployment Considerations . . . . . . . . . . . . . .  22
       4.3.6.  Security Considerations . . . . . . . . . . . . . . .  23
   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
     1.1.  Network Address Translators . . . . . . . . . . . . . . .   5
     1.2.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . .   6
     1.3.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   7
   2.  Detecting the Loss of NAT Mappings  . . . . . . . . . . . . .   8
   3.  Requirements on Solutions . . . . . . . . . . . . . . . . . .   9
   4.  NAT-Traversal Techniques  . . . . . . . . . . . . . . . . . .  10
     4.1.  Stand-Alone STUN  . . . . . . . . . . . . . . . . . . . .  11
       4.1.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  11
       4.1.2.  Using STUN to Traverse NAT without Server
               Modifications . . . . . . . . . . . . . . . . . . . .  11
       4.1.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  14
       4.1.4.  Deployment Considerations . . . . . . . . . . . . . .  14
       4.1.5.  Security Considerations . . . . . . . . . . . . . . .  15
     4.2.  Server Embedded STUN  . . . . . . . . . . . . . . . . . .  16
       4.2.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  16
       4.2.2.  Embedding STUN in RTSP  . . . . . . . . . . . . . . .  16
       4.2.3.  Discussion on Co-located STUN Server  . . . . . . . .  17
       4.2.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  17
       4.2.5.  Deployment Considerations . . . . . . . . . . . . . .  18
       4.2.6.  Security Considerations . . . . . . . . . . . . . . .  19
     4.3.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  19
       4.3.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  19
       4.3.2.  Using ICE in RTSP . . . . . . . . . . . . . . . . . .  20
       4.3.3.  Implementation Burden of ICE  . . . . . . . . . . . .  21
       4.3.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  22
       4.3.5.  Deployment Considerations . . . . . . . . . . . . . .  22
       4.3.6.  Security Considerations . . . . . . . . . . . . . . .  23
     4.4.  Latching  . . . . . . . . . . . . . . . . . . . . . . . .  23
       4.4.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  23
       4.4.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  24
       4.4.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  25
       4.4.4.  Deployment Considerations . . . . . . . . . . . . . .  25
       4.4.5.  Security Considerations . . . . . . . . . . . . . . .  26
     4.5.  A Variation to Latching . . . . . . . . . . . . . . . . .  27
       4.5.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  27
       4.5.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  28
       4.5.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  28
       4.5.4.  Deployment Considerations . . . . . . . . . . . . . .  28
       4.5.5.  Security Considerations . . . . . . . . . . . . . . .  29
     4.6.  Three-Way Latching  . . . . . . . . . . . . . . . . . . .  29
       4.6.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  29
       4.6.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  29
       4.6.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  30
       4.6.4.  Deployment Considerations . . . . . . . . . . . . . .  30
       4.6.5.  Security Considerations . . . . . . . . . . . . . . .  30
     4.7.  Application Level Gateways  . . . . . . . . . . . . . . .  31
       4.7.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  31
       4.7.2.  Outline on How ALGs for RTSP Work . . . . . . . . . .  31
       4.7.3.  Deployment Considerations . . . . . . . . . . . . . .  32
       4.7.4.  Security Considerations . . . . . . . . . . . . . . .  33
     4.8.  TCP Tunneling . . . . . . . . . . . . . . . . . . . . . .  33
       4.8.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  33
       4.8.2.  Usage of TCP Tunneling in RTSP  . . . . . . . . . . .  34
       4.8.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  34
       4.8.4.  Deployment Considerations . . . . . . . . . . . . . .  34
       4.8.5.  Security Considerations . . . . . . . . . . . . . . .  35
     4.9.  Traversal Using Relays around NAT (TURN)  . . . . . . . .  35
       4.9.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  35
       4.9.2.  Usage of TURN with RTSP . . . . . . . . . . . . . . .  36
       4.9.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  37
       4.9.4.  Deployment Considerations . . . . . . . . . . . . . .  37
       4.9.5.  Security Considerations . . . . . . . . . . . . . . .  37
   5.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . .  38
   6.  Comparison of NAT Traversal Techniques  . . . . . . . . . . .  39
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  41
   8.  Informative References  . . . . . . . . . . . . . . . . . . .  42
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  45
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  46
     4.4.  Latching  . . . . . . . . . . . . . . . . . . . . . . . .  23
       4.4.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  23
       4.4.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  24
       4.4.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  25
       4.4.4.  Deployment Considerations . . . . . . . . . . . . . .  25
       4.4.5.  Security Considerations . . . . . . . . . . . . . . .  26
     4.5.  A Variation to Latching . . . . . . . . . . . . . . . . .  27
       4.5.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  27
       4.5.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  28
       4.5.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  28
       4.5.4.  Deployment Considerations . . . . . . . . . . . . . .  28
       4.5.5.  Security Considerations . . . . . . . . . . . . . . .  29
     4.6.  Three-Way Latching  . . . . . . . . . . . . . . . . . . .  29
       4.6.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  29
       4.6.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  29
       4.6.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  30
       4.6.4.  Deployment Considerations . . . . . . . . . . . . . .  30
       4.6.5.  Security Considerations . . . . . . . . . . . . . . .  30
     4.7.  Application Level Gateways  . . . . . . . . . . . . . . .  31
       4.7.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  31
       4.7.2.  Outline on How ALGs for RTSP Work . . . . . . . . . .  31
       4.7.3.  Deployment Considerations . . . . . . . . . . . . . .  32
       4.7.4.  Security Considerations . . . . . . . . . . . . . . .  33
     4.8.  TCP Tunneling . . . . . . . . . . . . . . . . . . . . . .  33
       4.8.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  33
       4.8.2.  Usage of TCP Tunneling in RTSP  . . . . . . . . . . .  34
       4.8.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  34
       4.8.4.  Deployment Considerations . . . . . . . . . . . . . .  34
       4.8.5.  Security Considerations . . . . . . . . . . . . . . .  35
     4.9.  Traversal Using Relays around NAT (TURN)  . . . . . . . .  35
       4.9.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  35
       4.9.2.  Usage of TURN with RTSP . . . . . . . . . . . . . . .  36
       4.9.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  37
       4.9.4.  Deployment Considerations . . . . . . . . . . . . . .  37
       4.9.5.  Security Considerations . . . . . . . . . . . . . . .  37
   5.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . .  38
   6.  Comparison of NAT Traversal Techniques  . . . . . . . . . . .  39
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  41
   8.  Informative References  . . . . . . . . . . . . . . . . . . .  42
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  45
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  46
1. Introduction
1. 介绍

Today there is a proliferating deployment of different types of Network Address Translator (NAT) boxes that in many cases only loosely follow standards [RFC3022] [RFC2663] [RFC3424] [RFC4787] [RFC5382]. NATs cause discontinuity in address realms [RFC3424]; therefore, an application protocol, such as the Real-Time Streaming Protocol (RTSP) [RFC2326] [RTSP], needs to deal with such discontinuities caused by NATs. The problem is that, being a media control protocol managing one or more media streams, RTSP carries network address and port information within its protocol messages. Because of this, even if RTSP itself, when carried over the Transmission Control Protocol (TCP) [RFC793], for example, is not blocked by NATs, its media streams may be blocked by NATs. This will occur unless special protocol provisions are added to support NAT traversal.


Like NATs, firewalls are also middleboxes that need to be considered. Firewalls help prevent unwanted traffic from getting in or out of the protected network. RTSP is designed such that a firewall can be configured to let RTSP-controlled media streams go through with limited implementation effort. The effort needed is to implement an Application Level Gateway (ALG) to interpret RTSP parameters. There is also a large class of firewalls, commonly home firewalls, that use a filtering behavior that appears to be the same as what NATs have. This type of firewall will be successfully traversed using the same solution as employed for NAT traversal, instead of relying on an RTSP ALG. Therefore, firewalls will also be discussed and some important differences highlighted.

与NAT一样,防火墙也是需要考虑的中间包。防火墙有助于防止不必要的流量进出受保护的网络。RTSP的设计使防火墙可以配置为允许RTSP控制的媒体流通过,而实施工作有限。所需的工作是实现应用程序级网关(ALG)来解释RTSP参数。还有一大类防火墙,通常是家庭防火墙,它们使用的过滤行为似乎与NAT相同。这种类型的防火墙将使用与NAT穿越相同的解决方案成功穿越,而不是依赖RTSP ALG。因此,还将讨论防火墙,并强调一些重要的区别。

This document describes several NAT traversal mechanisms for RTSP-controlled media streaming. Many of these NAT solutions fall into the category of "UNilateral Self-Address Fixing (UNSAF)" as defined in [RFC3424] and quoted below:


[UNSAF] is a process whereby some originating process attempts to determine or fix the address (and port) by which it is known - e.g. to be able to use address data in the protocol exchange, or to advertise a public address from which it will receive connections.


Following the guidelines spelled out in RFC 3424, we describe the required RTSP extensions for each method, transition strategies, and security concerns. The transition strategies are a discussion of how and if the method encourages a move towards not having any NATs on the path.

按照RFC 3424中阐述的指导原则,我们描述了每种方法所需的RTSP扩展、转换策略和安全问题。过渡策略讨论了该方法如何以及是否鼓励在路径上不使用任何NAT。

This document is capturing the evaluation done in the process to recommend firewall/NAT traversal methods for RTSP streaming servers based on [RFC2326] as well as the RTSP 2.0 core specification [RTSP]. The evaluation is focused on NAT traversal for the media streams carried over the User Datagram Protocol (UDP) [RFC768] with RTP [RFC3550] over UDP being the main case for such usage. The findings should be applicable to other protocols as long as they have similar properties.

本文档记录了为基于[RFC2326]以及RTSP 2.0核心规范[RTSP]的RTSP流式服务器推荐防火墙/NAT穿越方法的过程中所做的评估。评估的重点是通过用户数据报协议(UDP)[RFC768]承载的媒体流的NAT遍历,UDP上的RTP[RFC3550]是此类使用的主要情况。这些发现应该适用于其他协议,只要它们具有相似的属性。

At the time when the bulk of work on this document was done, a single level of NAT was the dominant deployment for NATs, and multiple levels of NATs, including Carrier-Grade NATs (CGNs), were not considered. Thus, any characterizations or findings may not be applicable in such scenarios, unless CGN or multiple levels of NATs are explicitly noted.


An RTSP NAT traversal mechanism based on Interactive Connectivity Establishment (ICE) is specified in "A Network Address Translator (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming Protocol (RTSP)" [RTSP-NAT].

“实时流协议(RTSP)控制的媒体的网络地址转换器(NAT)遍历机制”[RTSP-NAT]中规定了基于交互式连接建立(ICE)的RTSP NAT遍历机制。

1.1. Network Address Translators
1.1. 网络地址转换器

We begin by reviewing two quotes from Section 3 in "Network Address Translation (NAT) Behavioral Requirements for Unicast UDP" [RFC4787] concerning NATs and their terminology:


Readers are urged to refer to [RFC2663] for information on NAT taxonomy and terminology. Traditional NAT is the most common type of NAT device deployed. Readers may refer to [RFC3022] for detailed information on traditional NAT. Traditional NAT has two main varieties -- Basic NAT and Network Address/Port Translator (NAPT).


NAPT is by far the most commonly deployed NAT device. NAPT allows multiple internal hosts to share a single public IP address simultaneously. When an internal host opens an outgoing TCP or UDP session through a NAPT, the NAPT assigns the session a public IP address and port number, so that subsequent response packets from the external endpoint can be received by the NAPT, translated, and forwarded to the internal host. The effect is that the NAPT establishes a NAT session to translate the (private IP address, private port number) tuple to a (public IP address, public port number) tuple, and vice versa, for the duration of the session. An issue of relevance to peer-to-peer applications is how the NAT behaves when an internal host initiates multiple simultaneous sessions from a single (private IP, private port) endpoint to multiple distinct endpoints on the external network.


In this specification, the term "NAT" refers to both "Basic NAT" and "Network Address/Port Translator (NAPT)".


This document uses the term "Address and Port Mapping" as the translation between an external address and port and an internal address and port. Note that this is not the same as an "address binding" as defined in RFC 2663.

本文档使用术语“地址和端口映射”作为外部地址和端口与内部地址和端口之间的转换。请注意,这与RFC 2663中定义的“地址绑定”不同。

Note: In the above text, it would be more correct to use an external IP address instead of a public IP address. The external IP address is commonly a public one, but it might be of another type if the NAT's external side is in a private address domain.


In addition to the above quote, there exists a number of address and port mapping behaviors described in more detail in Section 4.1 of [RFC4787] that are highly relevant to the discussion in this document.


NATs also have a filtering behavior on traffic arriving on the external side. Such behavior affects how well different methods for NAT traversal works through these NATs. See Section 5 of [RFC4787] for more information on the different types of filtering that have been identified.


1.2. Firewalls
1.2. 防火墙

A firewall is a security gateway that enforces certain access control policies between two network administrative domains: a private domain (intranet) and an external domain, e.g., the Internet. Many organizations use firewalls to prevent intrusions and malicious attacks on computing resources in the private intranet [RFC2588].


A comparison between NAT and a firewall is given below:


1. A firewall sits at security enforcement/protection points, while NAT sits at borders between two address domains.

1. 防火墙位于安全强制/保护点,而NAT位于两个地址域之间的边界。

2. NAT does not in itself provide security, although some access control policies can be implemented using address translation schemes. The inherent filtering behaviors are commonly mistaken for real security policies.

2. NAT本身并不提供安全性,尽管一些访问控制策略可以使用地址转换方案实现。固有的过滤行为通常被误认为是真正的安全策略。

It should be noted that many NAT devices intended for Residential or Small Office, Home Office (SOHO) use include both NATs and firewall functionality.


1.3. Glossary
1.3. 术语汇编

Address-Dependent Mapping: The NAT reuses the port mapping for subsequent packets sent from the same internal IP address and port to the same external IP address, regardless of the external port; see [RFC4787].


Address and Port-Dependent Mapping: The NAT reuses the port mapping for subsequent packets sent from the same internal IP address and port to the same external IP address and port while the mapping is still active; see [RFC4787].


ALG: Application Level Gateway is an entity that can be embedded in a NAT or other middlebox to perform the application layer functions required for a particular protocol to traverse the NAT/middlebox.


Endpoint-Independent Mapping: The NAT reuses the port mapping for subsequent packets sent from the same internal IP address and port to any external IP address and port; see [RFC4787].


ICE: Interactive Connectivity Establishment; see [RFC5245].


DNS: Domain Name Service


DoS: Denial of Service


DDoS: Distributed Denial of Service


NAT: Network Address Translator; see [RFC3022].


NAPT: Network Address/Port Translator; see [RFC3022].


RTP: Real-Time Transport Protocol; see [RFC3550].


RTSP: Real-Time Streaming Protocol; see [RFC2326] and [RTSP].


RTT: Round Trip Times


SDP: Session Description Protocol; see [RFC4566].


SSRC: Synchronization source in RTP; see [RFC3550].


2. Detecting the Loss of NAT Mappings
2. 检测NAT映射的丢失

Several NAT traversal techniques in the next chapter make use of the fact that the NAT UDP mapping's external address and port can be discovered. This information is then utilized to traverse the NAT box. However, any such information is only good while the mapping is still valid. As the IAB's UNSAF document [RFC3424] points out, the mapping can either timeout or change its properties. It is therefore important for the NAT traversal solutions to handle the loss or change of NAT mappings, according to RFC 3424.

下一章中的几种NAT遍历技术利用了可以发现NAT UDP映射的外部地址和端口这一事实。然后利用该信息遍历NAT框。然而,只有当映射仍然有效时,任何此类信息才是好的。正如IAB的UNSAF文档[RFC3424]所指出的,映射可以超时,也可以更改其属性。因此,根据RFC 3424,NAT穿越解决方案处理NAT映射的丢失或更改非常重要。

First, since NATs may also dynamically reclaim or readjust address/ port translations, "keep-alive" and periodic repolling may be required according to RFC 3424. Second, it is possible to detect and recover from the situation where the mapping has been changed or removed. The loss of a mapping can be detected when no traffic arrives for a while. Below we will give some recommendations on how to detect the loss of NAT mappings when using RTP/RTCP under RTSP control.

首先,由于NAT还可以动态地回收或重新调整地址/端口转换,根据RFC 3424,可能需要“保持活动”和定期重新填充。其次,可以检测映射已更改或删除的情况并从中恢复。当一段时间内没有流量到达时,可以检测到映射丢失。下面我们将给出一些关于如何在RTSP控制下使用RTP/RTCP时检测NAT映射丢失的建议。

An RTP session normally has both RTP and RTCP streams. The loss of an RTP mapping can only be detected when expected traffic does not arrive. If a client does not receive media data within a few seconds after having received the "200 OK" response to an RTSP PLAY request that starts the media delivery, it may be the result of a middlebox blocking the traffic. However, for a receiver to be more certain to detect the case where no RTP traffic was delivered due to NAT trouble, one should monitor the RTCP Sender reports if they are received and not also blocked. The sender report carries a field telling how many packets the server has sent. If that has increased and no RTP packets have arrived for a few seconds, it is likely the mapping for the RTP stream has been removed.

RTP会话通常同时具有RTP和RTCP流。RTP映射的丢失只能在预期流量未到达时检测到。如果客户端在收到RTSP播放请求的“200 OK”响应(启动媒体传送)后几秒钟内未接收到媒体数据,则可能是由于中间盒阻塞了通信量。然而,为了让接收方更确定地检测到由于NAT故障而没有传递RTP通信量的情况,应该监控RTCP发送方报告,如果它们被接收并且没有被阻止。sender报告包含一个字段,告诉服务器已发送多少数据包。如果这增加了,并且几秒钟内没有RTP数据包到达,则RTP流的映射可能已被删除。

The loss of mapping for RTCP is simpler to detect. RTCP is normally sent periodically in each direction, even during the RTSP ready state. If RTCP packets are missing for several RTCP intervals, the mapping is likely lost. Note that if neither RTCP packets nor RTSP messages are received by the RTSP server for a while (default 60 seconds), the RTSP server has the option to delete the corresponding RTP session, SSRC and RTSP session ID, because either the client can not get through a middlebox NAT/firewall, or the client is malfunctioning.


3. Requirements on Solutions
3. 对解决方案的要求

This section considers the set of requirements for the evaluation of RTSP NAT traversal solutions.

本节考虑评估RTSP NAT穿越解决方案的一系列要求。

RTSP is a client-server protocol. Typically, service providers deploy RTSP servers on the Internet or otherwise reachable address realm. However, there are use cases where the reverse is true: RTSP clients are connecting from any address realm to RTSP servers behind NATs, e.g., in a home. This is the case, for instance, when home surveillance cameras running as RTSP servers intend to stream video to cell phone users in the public address realm through a home NAT. In terms of requirements, the primary issue to solve is the RTSP NAT traversal problem for RTSP servers deployed in a network where the server is on the external side of any NAT, i.e., the server is not behind a NAT. The server behind a NAT is desirable but of much lower priority.

RTSP是一种客户机-服务器协议。通常,服务提供商在Internet或其他可访问的地址域上部署RTSP服务器。然而,也有相反的情况:RTSP客户端从任何地址域连接到NAT后面的RTSP服务器,例如在家中。例如,当作为RTSP服务器运行的家庭监控摄像机打算通过家庭NAT向公共广播领域的手机用户传输视频时,就是这种情况。就需求而言,要解决的主要问题是RTSP服务器的RTSP NAT遍历问题,该服务器部署在任何NAT的外部,即服务器不在NAT后面的网络中。NAT背后的服务器是理想的,但优先级要低得多。

Important considerations for any NAT traversal technique are whether any protocol modifications are needed and where the implementation burden resides (e.g., server, client, or middlebox). If the incentive to get RTSP to work over a NAT is sufficient, it will motivate the owner of the server, client, or middlebox to update, configure, or otherwise perform changes to the device and its software in order to support NAT traversal. Thus, the questions of who this burden falls on and how big it is are highly relevant.


The list of feature requirements for RTSP NAT solutions are given below:

RTSP NAT解决方案的功能要求列表如下:

1. Must work for all flavors of NATs, including NATs with Address and Port-Dependent Filtering.

1. 必须适用于所有类型的NAT,包括带有地址和端口相关过滤的NAT。

2. Must work for firewalls (subject to pertinent firewall administrative policies), including those with ALGs.

2. 必须适用于防火墙(根据相关防火墙管理政策),包括使用ALG的防火墙。

3. Should have minimal impact on clients not behind NATs and that are not dual hosted. RTSP dual hosting means that the RTSP signaling protocol and the media protocol (e.g., RTP) are implemented on different computers with different IP addresses.

3. 应该对不在NAT后面并且不是双主机的客户端产生最小的影响。RTSP双重托管意味着RTSP信令协议和媒体协议(如RTP)在具有不同IP地址的不同计算机上实现。

* For instance, no extra protocol RTT before arrival of media.

* 例如,在介质到达之前没有额外的协议RTT。

4. Should be simple to use/implement/administer so people actually turn them on.

4. 应该易于使用/实施/管理,以便人们真正打开它们。

* Discovery of the address(es) assigned by NAT should happen automatically, if possible.

* 如果可能,NAT分配的地址的发现应该自动进行。

5. Should authenticate dual-hosted client's media transport receiver to prevent usage of RTSP servers for DDoS attacks.

5. 应验证双主机客户端的媒体传输接收器,以防止使用RTSP服务器进行DDoS攻击。

The last requirement addresses the Distributed Denial-of-Service (DDoS) threat, which relates to NAT traversal as explained below.


During NAT traversal, when the RTSP server determines the media destination (address and port) for the client, the result may be that the IP address of the RTP receiver host is different than the IP address of the RTSP client host. This poses a DDoS threat that has significant amplification potentials because the RTP media streams in general consist of a large number of IP packets. DDoS attacks can occur if the attacker can fake the messages in the NAT traversal mechanism to trick the RTSP server into believing that the client's RTP receiver is located on a host to be attacked. For example, user A may use his RTSP client to direct the RTSP server to send video RTP streams to in order to degrade the services provided by


Note that a simple mitigation is for the RTSP server to disallow the cases where the client's RTP receiver has a different IP address than that of the RTSP client. This is recommended behavior in RTSP 2.0 unless other solutions to prevent this attack are present; see Section 21.2.1 in [RTSP]. With the increased deployment of NAT middleboxes by operators, i.e., CGN, the reuse of an IP address on the NAT's external side by many customers reduces the protection provided. Also in some applications (e.g., centralized conferencing), dual-hosted RTSP/RTP clients have valid use cases. The key is how to authenticate the messages exchanged during the NAT traversal process.

请注意,一个简单的缓解措施是RTSP服务器不允许客户端的RTP接收器的IP地址与RTSP客户端的IP地址不同的情况。这是RTSP 2.0中的推荐行为,除非存在防止此攻击的其他解决方案;参见[RTSP]中的第21.2.1节。随着运营商(即CGN)增加NAT中间盒的部署,许多客户在NAT的外部重新使用IP地址,减少了提供的保护。此外,在某些应用程序中(例如,集中式会议),双主机RTSP/RTP客户端具有有效的用例。关键是如何验证NAT遍历过程中交换的消息。

4. NAT-Traversal Techniques
4. NAT穿越技术

There exists a number of potential NAT traversal techniques that can be used to allow RTSP to traverse NATs. They have different features and are applicable to different topologies; their costs are also different. They also vary in security levels. In the following sections, each technique is outlined with discussions on the corresponding advantages and disadvantages.


The survey of traversal techniques was done prior to 2007 and is based on what was available then. This section includes NAT traversal techniques that have not been formally specified anywhere else. This document may be the only publicly available specification of some of the NAT traversal techniques. However, that is not a real barrier against doing an evaluation of the NAT traversal techniques. Some techniques used as part of some of the traversal solutions have been recommended against or are no longer possible due to the outcome


of standardization work or their failure to progress within IETF after the initial evaluation in this document. For example, RTP No-Op [RTP-NO-OP] was a proposed RTP payload format that failed to be specified; thus, it is not available for use today. In each such case, the missing parts will be noted and some basic reasons will be given.

在本文件中进行初步评估后,标准化工作或其未能在IETF内取得进展。例如,RTP No Op[RTP-No-Op]是一种未指定的拟议RTP有效负载格式;因此,它今天无法使用。在每种情况下,都会注意到缺失的部分,并给出一些基本原因。

4.1. Stand-Alone STUN
4.1. 单独昏迷
4.1.1. Introduction
4.1.1. 介绍

Session Traversal Utilities for NAT (STUN) [RFC5389] is a standardized protocol that allows a client to use secure means to discover the presence of a NAT between itself and the STUN server. The client uses the STUN server to discover the address and port mappings assigned by the NAT. Then using the knowledge of these NAT mappings, it uses the external addresses to directly connect to the independent RTSP server. However, this is only possible if the NAT address and port mapping behavior is such that the STUN server and RTSP server will see the same external address and port for the same internal address and port.


STUN is a client-server protocol. The STUN client sends a request to a STUN server and the server returns a response. There are two types of STUN messages -- Binding Requests and Indications. Binding Requests are used when determining a client's external address and soliciting a response from the STUN server with the seen address. Indications are used by the client for keep-alive messages towards the server and requires no response from the server.


The first version of STUN [RFC3489] included categorization and parameterization of NATs. This was abandoned in the updated version [RFC5389] due to it being unreliable and brittle. This particular traversal method uses the removed functionality described in RFC 3489 to detect the NAT type to give an early failure indication when the NAT is showing the behavior that this method can't support. This method also suggests using the RTP No-Op payload format [RTP-NO-OP] for keep-alives of the RTP traffic in the client-to-server direction. This can be replaced with another form of UDP packet as will be further discussed below.

STUN的第一个版本[RFC3489]包括NAT的分类和参数化。由于不可靠且易碎,更新版本[RFC5389]中放弃了该功能。此特定的遍历方法使用RFC 3489中描述的已删除功能来检测NAT类型,以便在NAT显示此方法无法支持的行为时给出早期故障指示。此方法还建议使用RTP No-Op有效负载格式[RTP-No-Op],以保持客户端到服务器方向上的RTP流量有效。这可以用另一种形式的UDP数据包代替,下面将进一步讨论。

4.1.2. Using STUN to Traverse NAT without Server Modifications
4.1.2. 使用STUN在不修改服务器的情况下遍历NAT

This section describes how a client can use STUN to traverse NATs to RTSP servers without requiring server modifications. Note that this method has limited applicability and requires the server to be available in the external/public address realm in regards to the client located behind a NAT(s).




o The server must be located in either a public address realm or the next-hop external address realm in regards to the client.

o 对于客户端,服务器必须位于公共地址域或下一跳外部地址域中。

o The client may only be located behind NATs that perform Endpoint-Independent or Address-Dependent Mappings (the STUN server and RTSP server on the same IP address). Clients behind NATs that do Address and Port-Dependent Mappings cannot use this method. See [RFC4787] for the full definition of these terms.

o 客户机只能位于执行端点无关或地址相关映射的NAT后面(同一IP地址上的STUN服务器和RTSP服务器)。NAT后面进行地址和端口相关映射的客户端不能使用此方法。有关这些术语的完整定义,请参见[RFC4787]。

o Based on the discontinued middlebox classification of the replaced STUN specification [RFC3489]; thus, it is brittle and unreliable.

o 基于替换的STUN规范[RFC3489]的中断的中间盒分类;因此,它是脆弱和不可靠的。



An RTSP client using RTP transport over UDP can use STUN to traverse a NAT(s) in the following way:


1. Use STUN to try to discover the type of NAT and the timeout period for any UDP mapping on the NAT. This is recommended to be performed in the background as soon as IP connectivity is established. If this is performed prior to establishing a streaming session, the delays in the session establishment will be reduced. If no NAT is detected, normal SETUP should be used.

1. 使用STUN尝试发现NAT的类型以及NAT上任何UDP映射的超时时间。建议在IP连接建立后立即在后台执行此操作。如果这是在建立流式会话之前执行的,则会话建立中的延迟将减少。如果未检测到NAT,则应使用正常设置。

2. The RTSP client determines the number of UDP ports needed by counting the number of needed media transport protocols sessions in the multimedia presentation. This information is available in the media description protocol, e.g., SDP [RFC4566]. For example, each RTP session will in general require two UDP ports: one for RTP, and one for RTCP.

2. RTSP客户端通过计算多媒体演示文稿中所需的媒体传输协议会话数来确定所需的UDP端口数。该信息在媒体描述协议中可用,例如SDP[RFC4566]。例如,每个RTP会话通常需要两个UDP端口:一个用于RTP,一个用于RTCP。

3. For each UDP port required, establish a mapping and discover the public/external IP address and port number with the help of the STUN server. A successful mapping looks like: client's local address/port <-> public address/port.

3. 对于所需的每个UDP端口,在STUN服务器的帮助下建立映射并发现公共/外部IP地址和端口号。成功的映射如下所示:客户端的本地地址/端口<->公共地址/端口。

4. Perform the RTSP SETUP for each media. In the Transport header, the following parameter should be included with the given values: "dest_addr" [RTSP] or "destination" + "client_port" [RFC2326] with the public/external IP address and port pair for both RTP and RTCP. To be certain that this works, servers must allow a client to set up the RTP stream on any port, not only even ports and with non-contiguous port numbers for RTP and RTCP. This requires the new feature provided in RTSP 2.0 [RTSP]. The server should respond with a Transport header containing an "src_addr"

4. 为每个介质执行RTSP设置。在传输标头中,以下参数应包含在给定值中:“dest_addr”[RTSP]或“destination”+“client_port”[RFC2326],其中包含RTP和RTCP的公共/外部IP地址和端口对。为了确保这一点,服务器必须允许客户机在任何端口上设置RTP流,而不仅仅是偶数端口以及RTP和RTCP的非连续端口号。这需要RTSP 2.0[RTSP]中提供的新功能。服务器应使用包含“src_addr”的传输头进行响应

or "source" + "server_port" parameters with the RTP and RTCP source IP address and port of the media stream.


5. To keep the mappings alive, the client should periodically send UDP traffic over all mappings needed for the session. For the mapping carrying RTCP traffic, the periodic RTCP traffic is likely enough. For mappings carrying RTP traffic and for mappings carrying RTCP packets at too low of a frequency, keep-alive messages should be sent.

5. 要使映射保持活动状态,客户端应定期通过会话所需的所有映射发送UDP通信。对于承载RTCP流量的映射,周期性RTCP流量可能就足够了。对于承载RTP流量的映射和承载RTCP数据包的映射,频率过低时,应发送保持活动的消息。

If a UDP mapping is lost, the above discovery process must be repeated. The media stream also needs to be SETUP again to change the transport parameters to the new ones. This will cause a glitch in media playback.


To allow UDP packets to arrive from the server to a client behind an Address-Dependent or Address and Port-Dependent Filtering NAT, the client must first send a UDP packet to establish the filtering state in the NAT. The client, before sending an RTSP PLAY request, must send a so-called hole-punching packet on each mapping to the IP address and port given as the server's source address and port. For a NAT that only is Address-Dependent Filtering, the hole-punching packet could be sent to the server's discard port (port number 9). For Address and Port-Dependent Filtering NATs, the hole-punching packet must go to the port used for sending UDP packets to the client. To be able to do that, the server needs to include the "src_addr" in the Transport header (which is the "source" transport parameter in RFC2326). Since UDP packets are inherently unreliable, to ensure that at least one UDP message passes the NAT, hole-punching packets should be retransmitted a reasonable number of times.


One could have used RTP No-Op packets [RTP-NO-OP] as hole-punching and keep-alive messages had they been defined. That would have ensured that the traffic would look like RTP and thus would likely have the least risk of being dropped by any firewall. The drawback of using RTP No-Op is that the payload type number must be dynamically assigned through RTSP first. Other options are STUN, an RTP packet without any payload, or a UDP packet without any payload. For RTCP it is most suitable to use correctly generated RTCP packets. In general, sending unsolicited traffic to the RTSP server may trigger security functions resulting in the blocking of the keep-alive messages or termination of the RTSP session itself.

如果定义了RTP No-Op数据包[RTP-No-Op],则可以将其用作穿孔和保持活动状态的消息。这将确保流量看起来像RTP,因此可能被任何防火墙丢弃的风险最小。使用RTP No Op的缺点是必须首先通过RTSP动态分配有效负载类型号。其他选项包括STUN、不带任何负载的RTP数据包或不带任何负载的UDP数据包。对于RTCP,最适合使用正确生成的RTCP数据包。通常,向RTSP服务器发送未经请求的通信量可能会触发安全功能,从而导致保持活动消息的阻塞或RTSP会话本身的终止。

This method is further brittle as it doesn't support Address and Port-Dependent Mappings. Thus, it proposes to use the old STUN methods to classify the NAT behavior, thus enabling early error indication. This is strictly not required but will lead to failures during setup when the NAT has the wrong behavior. This failure can


also occur if the NAT changes the properties of the existing mapping and filtering state or between the classification message exchange and the actual RTSP session setup, for example, due to load.


4.1.3. ALG Considerations
4.1.3. ALG注意事项

If a NAT supports RTSP ALG (Application Level Gateway) and is not aware of the STUN traversal option, service failure may happen, because a client discovers its NAT external IP address and port numbers and inserts them in its SETUP requests. When the RTSP ALG processes the SETUP request, it may change the destination and port number, resulting in unpredictable behavior. An ALG should not update address fields that contain addresses other than the NAT's internal address domain. In cases where the ALG modifies fields unnecessarily, two alternatives exist:

如果NAT支持RTSP ALG(应用程序级网关),并且不知道STUN遍历选项,则可能会发生服务故障,因为客户端会发现其NAT外部IP地址和端口号,并将其插入其设置请求中。当RTSP ALG处理设置请求时,它可能会更改目标和端口号,从而导致不可预测的行为。ALG不应更新包含NAT内部地址域以外地址的地址字段。如果ALG不必要地修改字段,则存在两种备选方案:

1. Use Transport Layer Security (TLS) to encrypt the data over the RTSP TCP connection to prevent the ALG from reading and modifying the RTSP messages.

1. 使用传输层安全性(TLS)通过RTSP TCP连接加密数据,以防止ALG读取和修改RTSP消息。

2. Turn off the STUN-based NAT traversal mechanism.

2. 关闭基于STUN的NAT遍历机制。

As it may be difficult to determine why the failure occurs, the usage of TLS-protected RTSP message exchange at all times would avoid this issue.


4.1.4. Deployment Considerations
4.1.4. 部署注意事项

For the stand-alone usage of STUN, the following applies:




o STUN is a solution first used by applications based on SIP [RFC3261] (see Sections 1 and 2 of [RFC5389]). As shown above, with little or no changes, the RTSP application can reuse STUN as a NAT traversal solution, avoiding the pitfall of solving a problem twice.

o STUN是基于SIP[RFC3261]的应用程序首次使用的解决方案(见[RFC5389]第1节和第2节)。如上所示,RTSP应用程序只需很少或不做任何更改,就可以将STUN重用为NAT遍历解决方案,避免了两次解决问题的陷阱。

o Using STUN does not require RTSP server modifications, assuming it is a server that is compliant with RTSP 2.0; it only affects the client implementation.

o 使用STUN不需要修改RTSP服务器,假设它是符合RTSP 2.0的服务器;它只影响客户端实现。



o Requires a STUN server deployed in the same address domain as the server.

o 需要在与服务器相同的地址域中部署STUN服务器。

o Only works with NATs that perform Endpoint-Independent and Address-Dependent Mappings. Address and Port-Dependent Filtering NATs create some issues.

o 仅适用于执行端点独立和地址依赖映射的NAT。地址和端口相关的过滤NAT会产生一些问题。

o Brittle to NATs changing the properties of the NAT mapping and filtering.

o 更改NAT映射和过滤的属性会导致NAT的脆弱性。

o Does not work with Address and Port-Dependent Mapping NATs without server modifications.

o 在不修改服务器的情况下,无法使用与地址和端口相关的映射NAT。

o Will not work if a NAT uses multiple IP addresses, since RTSP servers generally require all media streams to use the same IP as used in the RTSP connection to prevent becoming a DDoS tool.

o 如果NAT使用多个IP地址,则将不起作用,因为RTSP服务器通常要求所有媒体流使用RTSP连接中使用的相同IP,以防止成为DDoS工具。

o Interaction problems exist when an RTSP-aware ALG interferes with the use of STUN for NAT traversal unless TLS-secured RTSP message exchange is used.

o 当RTSP感知ALG干扰STUN用于NAT遍历时,存在交互问题,除非使用TLS安全的RTSP消息交换。

o Using STUN requires that RTSP servers and clients support the updated RTSP specification [RTSP], because it is no longer possible to guarantee that RTP and RTCP ports are adjacent to each other, as required by the "client_port" and "server_port" parameters in RFC 2326.

o 使用STUN要求RTSP服务器和客户端支持更新的RTSP规范[RTSP],因为无法再按照RFC 2326中“客户端端口”和“服务器端口”参数的要求保证RTP和RTCP端口彼此相邻。



The usage of STUN can be phased out gradually as the first step of a STUN-capable server or client should be to check the presence of NATs. The removal of STUN capability in the client implementations will have to wait until there is absolutely no need to use STUN.


4.1.5. Security Considerations
4.1.5. 安全考虑

To prevent the RTSP server from being used as Denial-of-Service (DoS) attack tools, the RTSP Transport header parameters "destination" and "dest_addr" are generally not allowed to point to any IP address other than the one the RTSP message originates from. The RTSP server is only prepared to make an exception to this rule when the client is trusted (e.g., through the use of a secure authentication process or through some secure method of challenging the destination to verify its willingness to accept the RTP traffic). Such a restriction means that STUN in general does not work for use cases where RTSP and media transport go to different addresses.


STUN combined with RTSP that is restricted by destination address has the same security properties as the core RTSP. It is protected from being used as a DoS attack tool unless the attacker has the ability to spoof the TCP connection carrying RTSP messages.


Using STUN's support for message authentication and the secure transport of RTSP messages, attackers cannot modify STUN responses or RTSP messages (TLS) to change the media destination. This protects against hijacking; however, as a client can be the initiator of an attack, these mechanisms cannot securely prevent RTSP servers from being used as DoS attack tools.


4.2. Server Embedded STUN
4.2. 服务器嵌入式STUN
4.2.1. Introduction
4.2.1. 介绍

This section describes an alternative to the stand-alone STUN usage in the previous section that has quite significantly different behavior.


4.2.2. Embedding STUN in RTSP
4.2.2. 在RTSP中嵌入STUN

This section outlines the adaptation and embedding of STUN within RTSP. This enables STUN to be used to traverse any type of NAT, including Address and Port-Dependent Mapping NATs. This would require RTSP-level protocol changes.


This NAT traversal solution has limitations:


1. It does not work if both the RTSP client and RTSP server are behind separate NATs.

1. 如果RTSP客户端和RTSP服务器都位于单独的NAT后面,则此功能不起作用。

2. The RTSP server may, for security reasons, refuse to send media streams to an IP that is different from the IP in the client RTSP requests.

2. 出于安全原因,RTSP服务器可能会拒绝向与客户端RTSP请求中的IP不同的IP发送媒体流。

Deviations from STUN as defined in RFC 5389:

与RFC 5389中定义的STUN的偏差:

1. The RTSP application must provision the client with an identity and shared secret to use in the STUN authentication;

1. RTSP应用程序必须为客户端提供身份和共享机密,以便在STUN身份验证中使用;

2. We require the STUN server to be co-located on the RTSP server's media source ports.

2. 我们要求STUN服务器位于RTSP服务器的媒体源端口上。

If the STUN server is co-located with the RTSP server's media source port, an RTSP client using RTP transport over UDP can use STUN to traverse ALL types of NATs. In the case of Address and Port-Dependent Mapping NATs, the party on the inside of the NAT must initiate UDP traffic. The STUN Binding Request, being a UDP packet itself, can serve as the traffic initiating packet. Subsequently, both the STUN Binding Response packets and the RTP/RTCP packets can traverse the NAT, regardless of whether the RTSP server or the RTSP client is behind NAT (however, only one of them can be behind a NAT).


Likewise, if an RTSP server is behind a NAT, then an embedded STUN server must be co-located on the RTSP client's RTCP port. Also, it will become the client that needs to disclose his destination address rather than the server, so the server can correctly determine its NAT external source address for the media streams. In this case, we assume that the client has some means of establishing a TCP connection to the RTSP server behind NAT so as to exchange RTSP messages with the RTSP server, potentially using a proxy or static rules.


To minimize delay, we require that the RTSP server supporting this option must inform the client about the RTP and RTCP ports from where the server will send out RTP and RTCP packets, respectively. This can be done by using the "server_port" parameter in RFC 2326 and the "src_addr" parameter in [RTSP]. Both are in the RTSP Transport header. But in general, this strategy will require that one first does one SETUP request per media to learn the server ports, then perform the STUN checks, followed by a subsequent SETUP to change the client port and destination address to what was learned during the STUN checks.

为了最大限度地减少延迟,我们要求支持此选项的RTSP服务器必须通知客户端服务器将分别从中发送RTP和RTCP数据包的RTP和RTCP端口。这可以通过使用RFC 2326中的“server_port”参数和[RTSP]中的“src_addr”参数来实现。两者都位于RTSP传输标头中。但通常情况下,此策略需要先对每个介质执行一次设置请求以了解服务器端口,然后执行STUN检查,然后执行后续设置以将客户端端口和目标地址更改为STUN检查期间了解到的内容。

To be certain that RTCP works correctly, the RTSP endpoint (server or client) will be required to send and receive RTCP packets from the same port.


4.2.3. Discussion on Co-located STUN Server
4.2.3. 同位STUN服务器的探讨

In order to use STUN to traverse Address and Port-Dependent Filtering or Mapping NATs, the STUN server needs to be co-located with the streaming server media output ports. This creates a demultiplexing problem: we must be able to differentiate a STUN packet from a media packet. This will be done based on heuristics. The existing STUN heuristics is the first byte in the packet and the Magic Cookie field (added in RFC 5389), which works fine between STUN and RTP or RTCP where the first byte happens to be different. Thanks to the Magic Cookie field, it is unlikely that other protocols would be mistaken for a STUN packet, but this is not assured. For more discussion of this, please see Section 5.1.2 of [RFC5764].

为了使用STUN遍历地址和端口相关的过滤或映射NAT,STUN服务器需要与流媒体服务器的媒体输出端口位于同一位置。这就产生了一个解复用问题:我们必须能够区分STUN数据包和媒体数据包。这将基于启发式进行。现有的STUN启发式算法是数据包中的第一个字节和Magic Cookie字段(添加在RFC 5389中),该字段在STUN和RTP或RTCP之间工作良好,其中第一个字节恰好不同。多亏了Magic Cookie字段,其他协议不太可能被误认为是STUN数据包,但这一点并不确定。有关这方面的更多讨论,请参见[RFC5764]第5.1.2节。

4.2.4. ALG Considerations
4.2.4. ALG注意事项

The same ALG traversal considerations as for stand-alone STUN applies (Section 4.1.3).


4.2.5. Deployment Considerations
4.2.5. 部署注意事项

For the "Embedded STUN" method the following applies:




o STUN is a solution first used by SIP applications. As shown above, with little or no changes, the RTSP application can reuse STUN as a NAT traversal solution, avoiding the pitfall of solving a problem twice.

o STUN是SIP应用程序首次使用的解决方案。如上所示,RTSP应用程序只需很少或不做任何更改,就可以将STUN重用为NAT遍历解决方案,避免了两次解决问题的陷阱。

o STUN has built-in message authentication features, which makes it more secure against hijacking attacks. See the next section for an in-depth security discussion.

o STUN具有内置的消息身份验证功能,这使它更安全地抵御劫持攻击。有关深入的安全性讨论,请参见下一节。

o This solution works as long as there is only one RTSP endpoint in the private address realm, regardless of the NAT's type. There may even be multiple NATs (see Figure 1 in [RFC5389]).

o 只要私有地址域中只有一个RTSP端点,无论NAT的类型如何,此解决方案都可以工作。甚至可能有多个NAT(参见[RFC5389]中的图1)。

o Compared to other UDP-based NAT traversal methods in this document, STUN requires little new protocol development (since STUN is already an IETF standard), and most likely less implementation effort, since open source STUN server and client implementations are available [STUN-IMPL] [PJNATH].

o 与本文档中其他基于UDP的NAT遍历方法相比,STUN几乎不需要新的协议开发(因为STUN已经是IETF标准),而且很可能需要更少的实现工作,因为可以使用开源STUN服务器和客户端实现[STUN-IMPL][PJNAH]。



o Some extensions to the RTSP core protocol, likely signaled by RTSP feature tags, must be introduced.

o 必须引入RTSP核心协议的一些扩展,这些扩展可能由RTSP功能标签发出信号。

o Requires an embedded STUN server to be co-located on each of the RTSP server's media protocol's ports (e.g., RTP and RTCP ports), which means more processing is required to demultiplex STUN packets from media packets. For example, the demultiplexer must be able to differentiate an RTCP RR packet from a STUN packet and forward the former to the streaming server and the latter to the STUN server.

o 要求嵌入式STUN服务器位于RTSP服务器的每个媒体协议端口(例如RTP和RTCP端口)上,这意味着需要进行更多处理才能将STUN数据包与媒体数据包解复用。例如,解复用器必须能够区分RTCP RR数据包和STUN数据包,并将前者转发给流媒体服务器,后者转发给STUN服务器。

o Does not support use cases that require the RTSP connection and the media reception to happen at different addresses, unless the server's security policy is relaxed.

o 不支持要求RTSP连接和媒体接收在不同地址发生的用例,除非服务器的安全策略放宽。

o Interaction problems exist when an RTSP ALG is not aware of STUN unless TLS is used to protect the RTSP messages.

o 当RTSP ALG不知道STUN时,存在交互问题,除非使用TLS保护RTSP消息。

o Using STUN requires that RTSP servers and clients support the updated RTSP specification [RTSP], and they both agree to support the NAT traversal feature.

o 使用STUN需要RTSP服务器和客户端支持更新的RTSP规范[RTSP],并且它们都同意支持NAT遍历功能。

o Increases the setup delay with at least the amount of time it takes to perform STUN message exchanges. Most likely an extra SETUP sequence will be required.

o 增加设置延迟,至少增加执行STUN消息交换所需的时间。很可能需要额外的设置顺序。



The usage of STUN can be phased out gradually as the first step of a STUN-capable machine can be used to check the presence of NATs for the presently used network connection. The removal of STUN capability in the client implementations will have to wait until there is absolutely no need to use STUN, i.e., no NATs or firewalls.


4.2.6. Security Considerations
4.2.6. 安全考虑

See Stand-Alone STUN (Section 4.1.5).


4.3. ICE
4.3. 冰
4.3.1. Introduction
4.3.1. 介绍

Interactive Connectivity Establishment (ICE) [RFC5245] is a methodology for NAT traversal that has been developed for SIP using SDP offer/answer. The basic idea is to try, in a staggered parallel fashion, all possible connection addresses in which an endpoint may be reached. This allows the endpoint to use the best available UDP "connection" (meaning two UDP endpoints capable of reaching each other). The methodology has very nice properties in that basically all NAT topologies are possible to traverse.


Here is how ICE works at a high level. Endpoint A collects all possible addresses that can be used, including local IP addresses, STUN-derived addresses, Traversal Using Relay NAT (TURN) addresses, etc. On each local port that any of these address and port pairs lead to, a STUN server is installed. This STUN server only accepts STUN requests using the correct authentication through the use of a username and password.


Endpoint A then sends a request to establish connectivity with endpoint B, which includes all possible "destinations" [RFC5245] to get the media through to A. Note that each of A's local address/port pairs (host candidates and server reflexive base) has a co-located STUN server. B in turn provides A with all its possible destinations for the different media streams. A and B then uses a STUN client to try to reach all the address and port pairs specified by A from its corresponding destination ports. The destinations for which the STUN requests successfully complete are then indicated and one is selected.


If B fails to get any STUN response from A, all hope is not lost. Certain NAT topologies require multiple tries from both ends before successful connectivity is accomplished; therefore, requests are retransmitted multiple times. The STUN requests may also result in more connectivity alternatives (destinations) being discovered and conveyed in the STUN responses.


4.3.2. Using ICE in RTSP
4.3.2. 冰在RTSP中的应用

The usage of ICE for RTSP requires that both client and server be updated to include the ICE functionality. If both parties implement the necessary functionality, the following steps could provide ICE support for RTSP.

使用ICE for RTSP需要更新客户端和服务器以包含ICE功能。如果双方都实现了必要的功能,以下步骤可以为RTSP提供ICE支持。

This assumes that it is possible to establish a TCP connection for the RTSP messages between the client and the server. This is not trivial in scenarios where the server is located behind a NAT, and may require some TCP ports be opened, or proxies are deployed, etc.


The negotiation of ICE in RTSP of necessity will work different than in SIP with SDP offer/answer. The protocol interactions are different, and thus the possibilities for transfer of states are also somewhat different. The goal is also to avoid introducing extra delay in the setup process at least for when the server is not behind a NAT in regards to the client, and the client is either having an address in the same address domain or is behind the NAT(s), which can address the address domain of the server. This process is only intended to support PLAY mode, i.e., media traffic flows from server to client.


1. ICE usage begins in the SDP. The SDP for the service indicates that ICE is supported at the server. No candidates can be given here as that would not work with on demand, DNS load balancing, etc., which have the SDP indicate a resource on a server park rather than a specific machine.

1. 冰的使用从SDP开始。服务的SDP表示服务器支持ICE。此处不能给出任何候选项,因为这将不适用于按需、DNS负载平衡等,因为SDP指示服务器园区上的资源,而不是特定机器上的资源。

2. The client gathers addresses and puts together its candidates for each media stream indicated in the session description.

2. 客户端收集地址,并将会话描述中指示的每个媒体流的候选地址放在一起。

3. In each SETUP request, the client includes its candidates in an ICE-specific transport specification. For the server, this indicates the ICE support by the client. One candidate is the most prioritized candidate and here the prioritization for this address should be somewhat different compared to SIP. High-performance candidates are recommended rather than candidates with the highest likelihood of success, as it is more likely that a server is not behind a NAT compared to a SIP user agent.

3. 在每个设置请求中,客户机在ICE特定的传输规范中包含其候选者。对于服务器,这表示客户端对ICE的支持。一个候选者是优先级最高的候选者,与SIP相比,此地址的优先级应该有所不同。推荐高性能的候选者,而不是成功可能性最高的候选者,因为与SIP用户代理相比,服务器不在NAT后面的可能性更大。

4. The server responds to the SETUP (200 OK) for each media stream with its candidates. A server not behind a NAT usually only provides a single ICE candidate. Also, here one candidate is the server primary address.

4. 服务器对每个媒体流及其候选流的设置(200 OK)作出响应。不在NAT后面的服务器通常只提供一个ICE候选服务器。此外,这里还有一个候选服务器主地址。

5. The connectivity checks are performed. For the server, the connectivity checks from the server to the clients have an additional usage. They verify that there is someone willing to receive the media, thus preventing the server from unknowingly performing a DoS attack.

5. 将执行连接检查。对于服务器,从服务器到客户端的连接检查还有其他用途。它们验证是否有人愿意接收媒体,从而防止服务器在不知情的情况下执行DoS攻击。

6. Connectivity checks from the client promoting a candidate pair were successful. Thus, no further SETUP requests are necessary and processing can proceed with step 7. If an address other than the primary has been verified by the client to work, that address may then be promoted for usage in a SETUP request (go to step 7). If the checks for the available candidates failed and if further candidates have been derived during the connectivity checks, then those can be signaled in new candidate lines in a SETUP request updating the list (go to step 5).

6. 来自提升候选对的客户端的连接检查成功。因此,不需要进一步的设置请求,处理可以继续执行步骤7。如果客户机已验证主地址以外的地址有效,则该地址可在安装请求中升级使用(转至步骤7)。如果对可用候选项的检查失败,并且如果在连接检查期间派生了更多候选项,则可以在更新列表的设置请求中在新候选行中通知这些候选项(转至步骤5)。

7. Client issues the PLAY request. If the server also has completed its connectivity checks for the promoted candidate pair (based on the username as it may be derived addresses if the client was behind NAT), then it can directly answer 200 OK (go to step 8). If the connectivity check has not yet completed, it responds with a 1xx code to indicate that it is verifying the connectivity. If that fails within the set timeout, an error is reported back. The client needs to go back to step 6.

7. 客户端发出播放请求。如果服务器还完成了升级候选对的连接检查(基于用户名,因为如果客户端在NAT后面,它可能是派生地址),那么它可以直接回答200 OK(转到步骤8)。如果连接性检查尚未完成,它将以1xx代码响应,以指示正在验证连接性。如果在设置的超时内失败,则会报告错误。客户端需要返回到步骤6。

8. Process completed and media can be delivered. ICE candidates not used may be released.

8. 流程已完成,可以交付介质。未使用的ICE候选者可能会被释放。

To keep media paths alive, the client needs to periodically send data to the server. This will be realized with STUN. RTCP sent by the client should be able to keep RTCP open, but STUN will also be used for SIP based on the same motivations as for ICE.


4.3.3. Implementation Burden of ICE
4.3.3. ICE的实施负担

The usage of ICE will require that a number of new protocols and new RTSP/SDP features be implemented. This makes ICE the solution that has the largest impact on client and server implementations among all the NAT/firewall traversal methods in this document.


RTSP server implementation requirements are:


o STUN server features

o STUN服务器功能

o Limited STUN client features

o 有限的STUN客户端功能

o SDP generation with more parameters

o 具有更多参数的SDP生成

o RTSP error code for ICE extension

o ICE扩展的RTSP错误代码

RTSP client implementation requirements are:


o Limited STUN server features

o 有限的STUN服务器功能

o Limited STUN client features

o 有限的STUN客户端功能

o RTSP error code and ICE extension

o RTSP错误代码和ICE扩展

4.3.4. ALG Considerations
4.3.4. ALG注意事项

If there is an RTSP ALG that doesn't support the NAT traversal method, it may interfere with the NAT traversal. As the usage of ICE for the traversal manifests itself in the RTSP message primarily as a new transport specification, an ALG that passes through unknown will not prevent the traversal. An ALG that discards unknown specifications will, however, prevent the NAT traversal. These issues can be avoided by preventing the ALG to interfere with the signaling by using TLS for the RTSP message transport.

如果存在不支持NAT遍历方法的RTSP ALG,它可能会干扰NAT遍历。由于用于遍历的ICE主要作为新的传输规范在RTSP消息中显示,因此通过未知的ALG不会阻止遍历。然而,丢弃未知规范的ALG将阻止NAT遍历。通过使用TLS进行RTSP消息传输,防止ALG干扰信令,可以避免这些问题。

An ALG that supports this traversal method can, on the most basic level, just pass the transport specifications through. ALGs in NATs and firewalls could use the ICE candidates to establish a filtering state that would allow incoming STUN messages prior to any outgoing hole-punching packets, and in that way it could speed up the connectivity checks and reduce the risk of failures.


4.3.5. Deployment Considerations
4.3.5. 部署注意事项



o Solves NAT connectivity discovery for basically all cases as long as a TCP connection between the client and server can be established. This includes servers behind NATs. (Note that a proxy between address domains may be required to get TCP through.)

o 只要可以在客户端和服务器之间建立TCP连接,就基本上解决了所有情况下的NAT连接发现问题。这包括NAT后面的服务器。(请注意,可能需要地址域之间的代理才能通过TCP。)

o Improves defenses against DDoS attacks, since a media-receiving client requires authentications via STUN on its media reception ports.

o 提高对DDoS攻击的防御能力,因为媒体接收客户端需要通过其媒体接收端口上的STUN进行身份验证。



o Increases the setup delay with at least the amount of time it takes for the server to perform its STUN requests.

o 增加设置延迟,至少增加服务器执行其STUN请求所需的时间。

o Assumes that it is possible to demultiplex between the packets of the media protocol and STUN packets. This is possible for RTP as discussed, for example, in Section 5.1.2 of [RFC5764].

o 假设可以在媒体协议的数据包和STUN数据包之间解复用。如[RFC5764]第5.1.2节所述,这对于RTP是可能的。

o Has a fairly high implementation burden put on both the RTSP server and client. However, several open source ICE implementations do exist, such as [NICE] and [PJNATH].

o RTSP服务器和客户端都有相当高的实现负担。然而,确实存在一些开源ICE实现,例如[NICE]和[PJNATH]。

4.3.6. Security Considerations
4.3.6. 安全考虑

One should review the Security Considerations section of ICE and STUN to understand that ICE contains some potential issues. However, these can be avoided by correctly using ICE in RTSP. An important factor is to secure the signaling, i.e., use TLS between the RTSP client and server. In fact ICE does help avoid the DDoS attack issue with RTSP substantially as it reduces the possibility for a DDoS using RTSP servers on attackers that are on path between the RTSP server and the target and capable of intercepting the STUN connectivity check packets and correctly sending a response to the server. The ICE connectivity checks with their random transaction IDs from the server to the client serves as a return-routability check and prevents off-path attackers to succeed with address spoofing. This is similar to Mobile IPv6's return routability procedure (Section 5.2.5 of [RFC6275]).


4.4. Latching
4.4. 闭锁
4.4.1. Introduction
4.4.1. 介绍

Latching is a NAT traversal solution that is based on requiring RTSP clients to send UDP packets to the server's media output ports. Conventionally, RTSP servers send RTP packets in one direction: from server to client. Latching is similar to connection-oriented traffic, where one side (e.g., the RTSP client) first "connects" by sending an RTP packet to the other side's RTP port; the recipient then replies to the originating IP and Port. This method is also referred to as "late binding". It requires that all RTP/RTCP transport be done symmetrically. This in effect requires Symmetric RTP [RFC4961]. Refer to [RFC7362] for a description of the Latching of SIP-negotiated media streams in Session Border Controllers.


Specifically, when the RTSP server receives the Latching packet (a.k.a. hole-punching packet, since it is used to punch a hole in the


firewall/NAT) from its client, it copies the source IP and Port number and uses them as the delivery address for media packets. By having the server send media traffic back the same way as the client's packets are sent to the server, address and port mappings will be honored. Therefore, this technique works for all types of NATs, given that the server is not behind a NAT. However, it does require server modifications. The format of the Latching packet will have to be defined.


Latching is very vulnerable to both hijacking and becoming a tool in DDoS attacks (see Security Considerations in [RFC7362]) because attackers can simply forge the source IP and Port of the Latching packet. The rule for restricting IP addresses to one of the signaling connections will need to be applied here also. However, that does not protect against hijacking from another client behind the same NAT. This can become a serious issue in deployments with CGNs.


4.4.2. Necessary RTSP Extensions
4.4.2. 必要的RTSP扩展

To support Latching, RTSP signaling must be extended to allow the RTSP client to indicate that it will use Latching. The client also needs to be able to signal its RTP SSRC to the server in its SETUP request. The RTP SSRC is used to establish some basic level of security against hijacking attacks or simply to avoid mis-association when multiple clients are behind the same NAT. Care must be taken in choosing clients' RTP SSRC. First, it must be unique within all the RTP sessions belonging to the same RTSP session. Second, if the RTSP server is sending out media packets to multiple clients from the same send port, the RTP SSRC needs to be unique among those clients' RTP sessions. Recognizing that there is a potential that RTP SSRC collisions may occur, the RTSP server must be able to signal to a client that a collision has occurred and that it wants the client to use a different RTP SSRC carried in the SETUP response or use unique ports per RTSP session. Using unique ports limits an RTSP server in the number of sessions it can simultaneously handle per interface IP addresses.

为了支持锁存,必须扩展RTSP信令,以允许RTSP客户端指示它将使用锁存。客户机还需要能够在其设置请求中向服务器发送其RTP SSRC信号。RTP SSRC用于建立一些基本的安全级别,以防劫持攻击,或者只是在多个客户端位于同一NAT后面时避免错误关联。在选择客户的RTP SSRC时必须小心。首先,它在属于同一RTSP会话的所有RTP会话中必须是唯一的。其次,如果RTSP服务器从同一发送端口向多个客户端发送媒体包,则RTP SSRC需要在这些客户端的RTP会话中是唯一的。认识到存在可能发生RTP SSRC冲突的可能性,RTSP服务器必须能够向客户端发出发生冲突的信号,并希望客户端使用安装响应中携带的不同RTP SSRC,或每个RTSP会话使用唯一端口。使用唯一端口限制RTSP服务器可以同时处理每个接口IP地址的会话数。

The Latching packet as discussed above should have a field that can contain a client and RTP session identifier to correctly associate the Latching packet with the correct context. If an RTP packet is to be used, there would be a benefit to using a well-defined RTP payload format for this purpose as the No-Op payload format proposed [RTP-NO-OP]. However, in the absence of such a specification, an RTP packet without a payload could be used. Using SSRC is beneficial because RTP and RTCP both would work as is. However, other packet formats could be used that carry the necessary identification of the context, and such a solution is discussed in Section 4.5.


4.4.3. ALG Considerations
4.4.3. ALG注意事项

An RTSP ALG not supporting this method could interfere with the methods used to indicate that Latching is to be done, as well as the SSRC signaling, thus preventing the method from working. However, if the RTSP ALG instead opens the corresponding pinholes and creates the necessary mapping in the NAT, traversal will still work. Securing the RTSP message transport using TLS will avoid this issue.

不支持此方法的RTSP ALG可能会干扰用于指示要进行锁存的方法以及SSRC信令,从而阻止该方法工作。但是,如果RTSP ALG打开相应的针孔,并在NAT中创建必要的映射,则遍历仍然有效。使用TLS保护RTSP消息传输将避免此问题。

An RTSP ALG that supports this traversal method can for basic functionality simply pass the related signaling parameters transparently. Due to the security considerations for Latching, there might exist a benefit for an RTSP ALG that will enable NAT traversal to negotiate with the path and turn off the Latching procedures when the ALG handles this. However, this opens up to failure modes when there are multiple levels of NAT and only one supports an RTSP ALG.

支持这种遍历方法的RTSP ALG可以为基本功能简单地透明地传递相关的信令参数。由于锁存的安全性考虑,RTSP ALG可能有一个好处,即当ALG处理此问题时,允许NAT遍历与路径协商并关闭锁存过程。但是,当存在多个NAT级别且只有一个支持RTSP ALG时,这会导致出现故障模式。

4.4.4. Deployment Considerations
4.4.4. 部署注意事项



o Works for all types of client-facing NATs (requirement 1 in Section 3).

o 适用于所有类型的面向客户的NAT(第3节中的要求1)。

o Has little interaction problems with any RTSP ALG changing the client's information in the Transport header.

o 与任何RTSP ALG在传输头中更改客户端信息时几乎没有交互问题。



o Requires modifications to both the RTSP server and client.

o 需要对RTSP服务器和客户端进行修改。

o Limited to working with servers that are not behind a NAT.

o 仅限于使用不在NAT后面的服务器。

o The format of the packet for "connection setup" (a.k.a Latching packet) is not defined.

o 未定义“连接设置”的数据包格式(也称为闭锁数据包)。

o SSRC management if RTP is used for Latching due to risk for mis-association of clients to RTSP sessions at the server if SSRC collision occurs.

o SSRC管理如果RTP用于锁存,因为如果发生SSRC冲突,客户端与服务器上的RTSP会话存在错误关联的风险。

o Has significant security considerations (See Section 4.4.5), due to the lack of a strong authentication mechanism and will need to use address restrictions.

o 由于缺乏强大的身份验证机制,具有重要的安全考虑(见第4.4.5节),需要使用地址限制。

4.4.5. Security Considerations
4.4.5. 安全考虑

Latching's major security issue is that RTP streams can be hijacked and directed towards any target that the attacker desires unless address restrictions are used. In the case of NATs with multiple clients on the inside of them, hijacking can still occur. This becomes a significant threat in the context of CGNs.


The most serious security problem is the deliberate attack with the use of an RTSP client and Latching. The attacker uses RTSP to set up a media session. Then it uses Latching with a spoofed source address of the intended target of the attack. There is no defense against this attack other than restricting the possible address a Latching packet can come from to the same address as the RTSP TCP connection is from. This prevents Latching to be used in use cases that require different addresses for media destination and signaling. Even allowing only a limited address range containing the signaling address from where Latching is allowed opens up a significant vulnerability as it is difficult to determine the address usage for the network the client connects from.

最严重的安全问题是使用RTSP客户端和锁存进行蓄意攻击。攻击者使用RTSP设置媒体会话。然后,它使用锁存来伪造攻击目标的源地址。除了将锁存数据包可能来自的地址限制为与RTSP TCP连接来自的地址相同之外,没有其他防御措施。这可以防止在需要不同地址的介质目的地和信令的用例中使用锁存。即使只允许包含允许锁定的信令地址的有限地址范围,也会打开一个严重的漏洞,因为很难确定客户端连接的网络的地址使用情况。

A hijack attack can also be performed in various ways. The basic attack is based on the ability to read the RTSP signaling packets in order to learn the address and port the server will send from and also the SSRC the client will use. Having this information, the attacker can send its own Latching packets containing the correct RTP SSRC to the correct address and port on the server. The RTSP server will then use the source IP and Port from the Latching packet as the destination for the media packets it sends.

劫持攻击也可以以各种方式进行。基本攻击基于读取RTSP信令包的能力,以便了解服务器将从中发送的地址和端口以及客户端将使用的SSRC。有了这些信息,攻击者可以将自己包含正确RTP SSRC的锁存数据包发送到服务器上的正确地址和端口。然后,RTSP服务器将使用锁存数据包的源IP和端口作为其发送的媒体数据包的目的地。

Another variation of this attack is for a man in the middle to modify the RTP Latching packet being sent by a client to the server by simply changing the source IP and Port to the target one desires to attack.


One can fend off the snooping-based attack by applying encryption to the RTSP signaling transport. However, if the attacker is a man in the middle modifying Latching packets, the attack is impossible to defend against other than through address restrictions. As a NAT rewrites the source IP and (possibly) port, this cannot be authenticated, but authentication is required in order to protect against this type of DoS attack.


Yet another issue is that these attacks also can be used to deny the client the service it desires from the RTSP server completely. The attacker modifies or originates its own Latching packets with a port


other than what the legit Latching packets use, which results in the media server sending the RTP/RTCP traffic to ports the client isn't listening for RTP/RTCP on.


The amount of random non-guessable material in the Latching packet determines how well Latching can fend off stream hijacking performed by parties that are off the client-to-server network path, i.e., it lacks the capability to see the client's Latching packets. The proposal above uses the 32-bit RTP SSRC field to this effect. Therefore, it is important that this field is derived with a non-predictable random number generator. It should not be possible by knowing the algorithm used and a couple of basic facts to derive what random number a certain client will use.

锁存数据包中随机不可猜测的材料的数量决定了锁存如何能够很好地抵御客户端到服务器网络路径之外的各方执行的流外劫持,即,它缺乏查看客户端锁存数据包的能力。上述方案为此使用32位RTP SSRC字段。因此,使用不可预测的随机数生成器导出此字段非常重要。如果知道使用的算法和两个基本事实,就不可能得出某个客户端将使用的随机数。

An attacker not knowing the SSRC but aware of which port numbers that a server sends from can deploy a brute-force attack on the server by testing a lot of different SSRCs until it finds a matching one. Therefore, a server could implement functionality that blocks packets to ports or from sources that receive or send multiple Latching packets with different invalid SSRCs, especially when they are coming from the same IP and Port. Note that this mitigation in itself opens up a new venue for DoS attacks against legit users trying to latch.


To improve the security against attackers, the amount of random material could be increased. To achieve a longer random tag while still using RTP and RTCP, it will be necessary to develop RTP and RTCP payload formats for carrying the random material.


4.5. A Variation to Latching
4.5. 闭锁的一种变体
4.5.1. Introduction
4.5.1. 介绍

Latching as described above requires the usage of a valid RTP format as the Latching packet, i.e., the first packet that the client sends to the server to establish a bidirectional transport flow for RTP streams. There is currently no appropriate RTP packet format for this purpose, although the RTP No-Op format was a proposal to fix the problem [RTP-NO-OP]; however, that work was abandoned. [RFC6263] discusses the implication of different types of packets as keep-alives for RTP, and its findings are very relevant to the format of the Latching packet.

如上所述的锁存要求使用有效的RTP格式作为锁存分组,即客户端发送到服务器以建立RTP流的双向传输流的第一个分组。目前没有适合此目的的RTP数据包格式,尽管RTP no-Op格式是解决问题的建议[RTP-no-Op];然而,这项工作被放弃了。[RFC6263]讨论了不同类型的数据包作为RTP保存的含义,其发现与闭锁数据包的格式非常相关。

Meanwhile, there have been NAT/firewall traversal techniques deployed in the wireless streaming market place that use non-RTP messages as Latching packets. This section describes a variant based on a subset of those solutions that alters the previously described Latching solution.


4.5.2. Necessary RTSP Extensions
4.5.2. 必要的RTSP扩展

In this variation of Latching, the Latching packet is a small UDP packet that does not contain an RTP header. In response to the client's Latching packet, the RTSP server sends back a similar Latching packet as a confirmation so the client can stop the so-called "connection phase" of this NAT traversal technique. Afterwards, the client only has to periodically send Latching packets as keep-alive messages for the NAT mappings.


The server listens on its RTP-media output port and tries to decode any received UDP packet as the Latching packet. This is valid since an RTSP server is not expecting RTP traffic from the RTSP client. Then, it can correlate the Latching packet with the RTSP client's session ID or the client's SSRC and record the NAT bindings accordingly. The server then sends a Latching packet as the response to the client.


The Latching packet can contain the SSRC to identify the RTP stream, and care must be taken if the packet is bigger than 12 bytes, ensuring that it is distinctively different from RTP packets, whose header size is 12 bytes.


RTSP signaling can be added to do the following:


1. Enable or disable such Latching message exchanges. When the firewall/NAT has an RTSP-aware ALG, it is possible to disable Latching message exchange and let the ALG work out the address and port mappings.

1. 启用或禁用此类锁定消息交换。当防火墙/NAT具有RTSP感知ALG时,可以禁用锁存消息交换,并让ALG计算地址和端口映射。

2. Configure the number of retries and the retry interval of the Latching message exchanges.

2. 配置锁定消息交换的重试次数和重试间隔。

4.5.3. ALG Considerations
4.5.3. ALG注意事项

See Latching ALG considerations in Section 4.4.3.


4.5.4. Deployment Considerations
4.5.4. 部署注意事项

This approach has the following advantages when compared with the Latching approach (Section 4.4):


1. There is no need to define an RTP payload format for firewall traversal; therefore, it is more simple to use, implement, and administer (requirement 4 in Section 3) than a Latching protocol, which must be defined.

1. 不需要为防火墙遍历定义RTP有效负载格式;因此,它的使用、实现和管理(第3节中的要求4)比必须定义的锁存协议更简单。

2. When properly defined, this kind of Latching packet exchange can also authenticate RTP receivers, to prevent hijacking attacks.

2. 如果定义正确,这种锁存数据包交换还可以对RTP接收器进行身份验证,以防止劫持攻击。

This approach has the following disadvantage when compared with the Latching approach:


1. The server's sender SSRC for the RTP stream or other session Identity information must be signaled in the RTSP's SETUP response, in the Transport header of the RTSP SETUP response.

1. RTP流或其他会话标识信息的服务器发送方SSRC必须在RTSP设置响应的RTSP设置响应的传输头中发出信号。

4.5.5. Security Considerations
4.5.5. 安全考虑

Compared to the security properties of Latching, this variant is slightly improved. First of all it allows for a larger random field in the Latching packets, which makes it more unlikely for an off-path attacker to succeed in a hijack attack. Second, the confirmation allows the client to know when Latching works and when it doesn't and thus when to restart the Latching process by updating the SSRC.


Still, the main security issue remaining is that the RTSP server can't know that the source address in the Latching packet was coming from an RTSP client wanting to receive media and not from one that likes to direct the media traffic to a DoS target.


4.6. Three-Way Latching
4.6. 三向闭锁
4.6.1. Introduction
4.6.1. 介绍

Three-Way Latching is an attempt to try to resolve the most significant security issues for both previously discussed variants of Latching. By adding a server request response exchange directly after the initial Latching, the server can verify that the target address present in the Latching packet is an active listener and confirm its desire to establish a media flow.


4.6.2. Necessary RTSP Extensions
4.6.2. 必要的RTSP扩展

Uses the same RTSP extensions as the Alternative Latching method (Section 4.5) uses. The extensions for this variant are only in the format and transmission of the Latching packets.


The client-to-server Latching packet is similar to the Alternative Latching (Section 4.5), i.e., a UDP packet with some session identifiers and a random value. When the server responds to the Latching packet with a Latching confirmation, it includes a random value (nonce) of its own in addition to echoing back the one the client sent. Then a third message is added to the exchange. The client acknowledges the reception of the Latching confirmation


message and echoes back the server's nonce, thus confirming that the Latched address goes to an RTSP client that initiated the Latching and is actually present at that address. The RTSP server will refuse to send any media until the Latching Acknowledgement has been received with a valid nonce.


4.6.3. ALG Considerations
4.6.3. ALG注意事项

See Latching ALG considerations in Section 4.4.3.


4.6.4. Deployment Considerations
4.6.4. 部署注意事项

A solution with a three-way handshake and its own Latching packets can be compared with the ICE-based solution (Section 4.3) and have the following differences:


o Only works for servers that are not behind a NAT.

o 仅适用于不支持NAT的服务器。

o May be simpler to implement due to the avoidance of the ICE prioritization and check-board mechanisms.

o 由于避免了ICE优先级划分和检查委员会机制,实施起来可能更简单。

However, a Three-Way Latching protocol is very similar to using STUN in both directions as a Latching and verification protocol. Using STUN would remove the need for implementing a new protocol.


4.6.5. Security Considerations
4.6.5. 安全考虑

Three-Way Latching is significantly more secure than its simpler versions discussed above. The client-to-server nonce, which is included in signaling and also can be bigger than the 32 bits of random data that the SSRC field supports, makes it very difficult for an off-path attacker to perform a DoS attack by diverting the media.


The client-to-server nonce and its echoing back does not protect against on-path attackers, including malicious clients. However, the server-to-client nonce and its echoing back prevents malicious clients to divert the media stream by spoofing the source address and port, as it can't echo back the nonce in these cases. This is similar to the Mobile IPv6 return routability procedure (Section 5.2.5 of [RFC6275]).


Three-Way Latching is really only vulnerable to an on-path attacker that is quite capable. First, the attacker can learn the client-to-server nonce either by intercepting the signaling or by modifying the source information (target destination) of a client's Latching packet. Second, it is also on-path between the server and target destination and can generate a response using the server's nonce. An


adversary that has these capabilities is commonly capable of causing significantly worse damage than this using other methods.


Three-Way Latching results in the server-to-client packet being bigger than the client-to-server packet, due to the inclusion of the server-to-client nonce in addition to the client-to-server nonce. Thus, an amplification effect does exist; however, to achieve this amplification effect, the attacker has to create a session state on the RTSP server. The RTSP server can also limit the number of responses it will generate before considering the Latching to be failed.


4.7. Application Level Gateways
4.7. 应用程序级网关
4.7.1. Introduction
4.7.1. 介绍

An ALG reads the application level messages and performs necessary changes to allow the protocol to work through the middlebox. However, this behavior has some problems in regards to RTSP:


1. It does not work when RTSP is used with end-to-end security. As the ALG can't inspect and change the application level messages, the protocol will fail due to the middlebox.

1. 当RTSP与端到端安全性一起使用时,它不起作用。由于ALG无法检查和更改应用程序级消息,协议将由于中间盒而失败。

2. ALGs need to be updated if extensions to the protocol are added. Due to deployment issues with changing ALGs, this may also break the end-to-end functionality of RTSP.

2. 如果添加了协议扩展,则需要更新ALG。由于ALG发生变化时存在部署问题,这也可能会破坏RTSP的端到端功能。

Due to the above reasons, it is not recommended to use an RTSP ALG in NATs. This is especially important for NATs targeted to home users and small office environments, since it is very hard to upgrade NATs deployed in SOHO environments.

由于上述原因,不建议在NAT中使用RTSP ALG。这对于针对家庭用户和小型办公室环境的NAT尤其重要,因为升级SOHO环境中部署的NAT非常困难。

4.7.2. Outline on How ALGs for RTSP Work
4.7.2. RTSP的ALG工作原理概述

In this section, we provide a step-by-step outline on how one could go about writing an ALG to enable RTSP to traverse a NAT.


1. Detect any SETUP request.

1. 检测任何设置请求。

2. Try to detect the usage of any of the NAT traversal methods that replace the address and port of the Transport header parameters "destination" or "dest_addr". If any of these methods are used, then the ALG should not change the address. Ways to detect that these methods are used are:

2. 尝试检测替换传输头参数“destination”或“dest_addr”的地址和端口的任何NAT遍历方法的使用情况。如果使用这些方法中的任何一种,则ALG不应更改地址。检测是否使用这些方法的方法有:

* For embedded STUN, it would be to watch for a feature tag, like "nat.stun", and to see if any of those exist in the "supported", "proxy-require", or "require" headers of the RTSP exchange.

* 对于嵌入式STUN,需要注意一个功能标签,如“nat.STUN”,并查看RTSP交换的“受支持”、“代理请求”或“请求”头中是否存在这些标签。

* For stand-alone STUN and TURN-based solutions: This can be detected by inspecting the "destination" or "dest_addr" parameter. If it contains either one of the NAT's external IP addresses or a public IP address, then such a solution is in use. However, if multiple NATs are used, this detection may fail. Remapping should only be done for addresses belonging to the NAT's own private address space.

* 对于独立的眩晕和转身解决方案:这可以通过检查“目的地”或“目的地地址”参数来检测。若它包含NAT的一个外部IP地址或一个公共IP地址,那个么这样的解决方案正在使用中。但是,如果使用多个NAT,此检测可能会失败。只应对属于NAT自己的专用地址空间的地址进行重新映射。

Otherwise, continue to the next step.


3. Create UDP mappings (client given IP and Port <-> external IP and Port) where needed for all possible transport specifications in the Transport header of the request found in (step 1). Enter the external address and port(s) of these mappings in the Transport header. Mappings shall be created with consecutive external port numbers starting on an even number for RTP for each media stream. Mappings should also be given a long timeout period, at least 5 minutes.

3. 创建UDP映射(客户端给定的IP和端口<->外部IP和端口),以便在(步骤1)中找到的请求的传输头中的所有可能的传输规范都需要该映射。在传输标头中输入这些映射的外部地址和端口。应使用连续的外部端口号创建映射,从每个媒体流的RTP偶数开始。映射也应该有一个长的超时时间,至少5分钟。

4. When the SETUP response is received from the server, the ALG may remove the unused UDP mappings, i.e., the ones not present in the Transport header. The session ID should also be bound to the UDP mappings part of that session.

4. 当从服务器接收到设置响应时,ALG可能会删除未使用的UDP映射,即传输头中不存在的映射。会话ID还应绑定到该会话的UDP映射部分。

5. If the SETUP response settles on RTP over TCP or RTP over RTSP as lower transport, do nothing: let TCP tunneling take care of NAT traversal. Otherwise, go to the next step.

5. 如果设置响应以RTP over TCP或RTP over RTSP作为较低的传输,则什么也不做:让TCP隧道处理NAT遍历。否则,请转至下一步。

6. The ALG should keep the UDP mappings belonging to the RTSP session as long as: an RTSP message with the session's ID has been sent in the last timeout interval, or a UDP message has been sent on any of the UDP mappings during the last timeout interval.

6. ALG应保留属于RTSP会话的UDP映射,只要:具有会话ID的RTSP消息已在上一个超时间隔内发送,或UDP消息已在上一个超时间隔内发送到任何UDP映射。

7. The ALG may remove a mapping as soon as a TEARDOWN response has been received for that media stream.

7. 一旦接收到针对该媒体流的拆卸响应,ALG就可以移除映射。

4.7.3. Deployment Considerations
4.7.3. 部署注意事项



o No impact on either client or server.

o 对客户端或服务器都没有影响。

o Can work for any type of NATs.

o 可用于任何类型的NAT。



o When deployed, they are hard to update to reflect protocol modifications and extensions. If not updated, they will break the functionality.

o 部署时,它们很难更新以反映协议修改和扩展。如果不更新,它们将破坏功能。

o When end-to-end security is used, the ALG functionality will fail.

o 使用端到端安全性时,ALG功能将失败。

o Can interfere with other types of traversal mechanisms, such as STUN.

o 可能会干扰其他类型的遍历机制,例如STUN。



An RTSP ALG will not be phased out in any automatic way. It must be removed, probably through the removal or update of the NAT it is associated with.

RTSP ALG不会以任何自动方式逐步淘汰。必须删除它,可能是通过删除或更新与其关联的NAT。

4.7.4. Security Considerations
4.7.4. 安全考虑

An ALG will not work with deployment of end-to-end RTSP signaling security; however, it will work with the hop-by-hop security method defined in Section 19.3 of RTSP 2.0 [RTSP]. Therefore, deployment of ALG may result in clients located behind NATs not using end-to-end security, or more likely the selection of a NAT traversal solution that allows for security.

ALG不支持端到端RTSP信令安全的部署;但是,它将使用RTSP 2.0[RTSP]第19.3节中定义的逐跳安全方法。因此,ALG的部署可能导致位于NAT后面的客户端不使用端到端安全性,或者更可能是选择考虑安全性的NAT穿越解决方案。

The creation of a UDP mapping based on the signaling message has some potential security implications. First of all, if the RTSP client releases its ports and another application is assigned these instead, it could receive RTP media as long as the mappings exist and the RTSP server has failed to be signaled or notice the lack of client response.


A NAT with RTSP ALG that assigns mappings based on SETUP requests could potentially become the victim of a resource exhaustion attack. If an attacker creates a lot of RTSP sessions, even without starting media transmission, this could exhaust the pool of available UDP ports on the NAT. Thus, only a limited number of UDP mappings should be allowed to be created by the RTSP ALG.

基于设置请求分配映射的带有RTSP ALG的NAT可能成为资源耗尽攻击的受害者。如果攻击者创建大量RTSP会话,即使没有启动媒体传输,也可能耗尽NAT上可用UDP端口池。因此,RTSP ALG只允许创建有限数量的UDP映射。

4.8. TCP Tunneling
4.8. TCP隧道
4.8.1. Introduction
4.8.1. 介绍

Using a TCP connection that is established from the client to the server ensures that the server can send data to the client. The connection opened from the private domain ensures that the server can send data back to the client. To send data originally intended to be


transported over UDP requires the TCP connection to support some type of framing of the media data packets. Using TCP also results in the client having to accept that real-time performance can be impacted. TCP's problem of ensuring timely delivery was one of the reasons why RTP was developed. Problems that arise with TCP are: head-of-line blocking, delay introduced by retransmissions, and a highly varying rate due to the congestion control algorithm. If a sufficient amount of buffering (several seconds) in the receiving client can be tolerated, then TCP will clearly work.


4.8.2. Usage of TCP Tunneling in RTSP
4.8.2. TCP隧道在RTSP中的应用

The RTSP core specification [RTSP] supports interleaving of media data on the TCP connection that carries RTSP signaling. See Section 14 in [RTSP] for how to perform this type of TCP tunneling. There also exists another way of transporting RTP over TCP, which is defined in Appendix C.2 in [RTSP]. For signaling and rules on how to establish the TCP connection in lieu of UDP, see Appendix C.2 in [RTSP]. This is based on the framing of RTP over the TCP connection as described in [RFC4571].


4.8.3. ALG Considerations
4.8.3. ALG注意事项

An RTSP ALG will face a different issue with TCP tunneling, at least the interleaved version. Now the full data stream can end up flowing through the ALG implementation. Thus, it is important that the ALG is efficient in dealing with the interleaved media data frames to avoid consuming to many resources and thus creating performance issues.

RTSP ALG将在TCP隧道中面临不同的问题,至少是交叉版本。现在,完整的数据流可以通过ALG实现结束。因此,重要的是ALG在处理交织媒体数据帧时是有效的,以避免消耗许多资源,从而产生性能问题。

The RTSP ALG can also affect the transport specifications that indicate that TCP tunneling can be done and its prioritization, including removing the transport specification, thus preventing TCP tunneling.

RTSP ALG还可以影响传输规范,该规范指示可以进行TCP隧道及其优先级,包括删除传输规范,从而防止TCP隧道。

4.8.4. Deployment Considerations
4.8.4. 部署注意事项



o Works through all types of NATs where the RTSP server is not NATed or is at least reachable like it was not.

o 适用于RTSP服务器未启用或至少可以访问的所有类型的NAT。



o Functionality needs to be implemented on both server and client.

o 功能需要在服务器和客户端上实现。

o Will not always meet multimedia stream's real-time requirements.

o 不能始终满足多媒体流的实时性要求。



The tunneling over RTSP's TCP connection is not planned to be phased out. It is intended to be a fallback mechanism and for usage when total media reliability is desired, even at the potential price of loss of real-time properties.


4.8.5. Security Considerations
4.8.5. 安全考虑

The TCP tunneling of RTP has no known significant security problems besides those already presented in the RTSP specification. It is difficult to get any amplification effect for DoS attacks due to TCP's flow control. The RTSP server's TCP socket, if independently used for media tunneling or only RTSP messages, can be used for a redirected syn attack. By spoofing the source address of any TCP init packets, the TCP SYNs from the server can be directed towards a target.

RTP的TCP隧道除了RTSP规范中已经提出的安全问题外,没有已知的重大安全问题。由于TCP的流量控制,DoS攻击很难得到任何放大效果。RTSP服务器的TCP套接字如果单独用于媒体隧道或仅用于RTSP消息,则可用于重定向syn攻击。通过欺骗任何TCP init数据包的源地址,可以将服务器上的TCP SYN定向到目标。

A possible security consideration, when session media data is interleaved with RTSP, would be the performance bottleneck when RTSP encryption is applied, since all session media data also needs to be encrypted.


4.9. Traversal Using Relays around NAT (TURN)
4.9. 使用NAT周围的继电器进行遍历(转弯)
4.9.1. Introduction
4.9.1. 介绍

TURN [RFC5766] is a protocol for setting up traffic relays that allow clients behind NATs and firewalls to receive incoming traffic for both UDP and TCP. These relays are controlled and have limited resources. They need to be allocated before usage. TURN allows a client to temporarily bind an address/port pair on the relay (TURN server) to its local source address/port pair, which is used to contact the TURN server. The TURN server will then forward packets between the two sides of the relay.


To prevent DoS attacks on either recipient, the packets forwarded are restricted to the specific source address. On the client side, it is restricted to the source setting up the allocation. On the external side, it is limited to the source address/port pair that have been given permission by the TURN client creating the allocation. Packets from any other source on this address will be discarded.


Using a TURN server makes it possible for an RTSP client to receive media streams from even an unmodified RTSP server. However, the problem is those RTSP servers most likely restrict media destinations to no other IP address than the one the RTSP message arrives from. This means that TURN could only be used if the server knows and


accepts that the IP belongs to a TURN server, and the TURN server can't be targeted at an unknown address. Alternatively, both the RTSP TCP connection as well as the RTP media is relayed through the same TURN server.

接受IP属于TURN服务器,并且TURN服务器不能以未知地址为目标。或者,RTSP TCP连接和RTP介质都通过同一TURN服务器中继。

4.9.2. Usage of TURN with RTSP
4.9.2. 与RTSP一起使用TURN

To use a TURN server for NAT traversal, the following steps should be performed.


1. The RTSP client connects with the RTSP server. The client retrieves the session description to determine the number of media streams. To avoid the issue of having the RTSP connection and media traffic from different addresses, the TCP connection must also be done through the same TURN server as the one in the next step. This will require the usage of TURN for TCP [RFC6062].

1. RTSP客户端与RTSP服务器连接。客户端检索会话描述以确定媒体流的数量。为了避免RTSP连接和来自不同地址的媒体流量问题,TCP连接也必须通过与下一步相同的TURN服务器完成。这将需要对TCP使用TURN[RFC6062]。

2. The client establishes the necessary bindings on the TURN server. It must choose the local RTP and RTCP ports that it desires to receive media packets. TURN supports requesting bindings of even port numbers and contiguous ranges.

2. 客户端在TURN服务器上建立必要的绑定。它必须选择希望接收媒体数据包的本地RTP和RTCP端口。TURN支持请求偶数端口号和连续范围的绑定。

3. The RTSP client uses the acquired address and port allocations in the RTSP SETUP request using the destination header.

3. RTSP客户端使用目标标头在RTSP设置请求中使用获取的地址和端口分配。

4. The RTSP server sends the SETUP reply, which must include the Transport header's "src_addr" parameter (source and port in RTSP 1.0). Note that the server is required to have a mechanism to verify that it is allowed to send media traffic to the given address unless TCP relaying of the RTSP messages also is performed.

4. RTSP服务器发送安装应答,该应答必须包括传输头的“src_addr”参数(RTSP 1.0中的源和端口)。请注意,服务器需要有一种机制来验证是否允许向给定地址发送媒体流量,除非同时执行RTSP消息的TCP中继。

5. The RTSP client uses the RTSP server's response to create TURN permissions for the server's media traffic.

5. RTSP客户端使用RTSP服务器的响应为服务器的媒体流量创建TURN权限。

6. The client requests that the server starts playing. The server starts sending media packets to the given destination address and ports.

6. 客户端请求服务器开始播放。服务器开始向给定的目标地址和端口发送媒体数据包。

7. Media packets arrive at the TURN server on the external port; if the packets match an established permission, the TURN server forwards the media packets to the RTSP client.

7. 媒体包到达外部端口上的TURN服务器;如果数据包与已建立的权限匹配,TURN服务器将媒体数据包转发给RTSP客户端。

8. If the client pauses and media is not sent for about 75% of the mapping timeout, the client should use TURN to refresh the bindings.

8. 如果在大约75%的映射超时时间内,客户端暂停并且未发送媒体,则客户端应使用TURN刷新绑定。

4.9.3. ALG Considerations
4.9.3. ALG注意事项

As the RTSP client inserts the address information of the TURN relay's external allocations in the SETUP messages, the ALG that replaces the address, without considering that the address does not belong to the internal address realm of the NAT, will prevent this mechanism from working. This can be prevented by securing the RTSP signaling.


4.9.4. Deployment Considerations
4.9.4. 部署注意事项



o Does not require any server modifications given that the server includes the "src_addr" header in the SETUP response.

o 不需要任何服务器修改,因为服务器在安装响应中包含“src_addr”头。

o Works for any type of NAT as long as the RTSP server has a reachable IP address that is not behind a NAT.

o 只要RTSP服务器有一个不在NAT后面的可访问IP地址,就可以用于任何类型的NAT。



o Requires another network element, namely the TURN server.

o 需要另一个网元,即TURN服务器。

o A TURN server for RTSP may not scale since the number of sessions it must forward is proportional to the number of client media sessions.

o RTSP的TURN服务器可能无法扩展,因为它必须转发的会话数与客户端媒体会话数成比例。

o The TURN server becomes a single point of failure.

o TURN服务器成为单点故障。

o Since TURN forwards media packets, as a necessity it introduces delay.

o 由于TURN转发媒体数据包,因此必然会引入延迟。

o An RTSP ALG may change the necessary destinations parameter. This will cause the media traffic to be sent to the wrong address.

o RTSP ALG可更改必要的目的地参数。这将导致媒体流量发送到错误的地址。



TURN is not intended to be phased out completely; see Section 19 of [RFC5766]. However, the usage of TURN could be reduced when the demand for having NAT traversal is reduced.


4.9.5. Security Considerations
4.9.5. 安全考虑

The TURN server can become part of a DoS attack towards any victim. To perform this attack, the attacker must be able to eavesdrop on the packets from the TURN server towards a target for the DoS attack. The attacker uses the TURN server to set up an RTSP session with media flows going through the TURN server. The attacker is in fact


creating TURN mappings towards a target by spoofing the source address of TURN requests. As the attacker will need the address of these mappings, he must be able to eavesdrop or intercept the TURN responses going from the TURN server to the target. Having these addresses, he can set up an RTSP session and start delivery of the media. The attacker must be able to create these mappings. The attacker in this case may be traced by the TURN username in the mapping requests.


This attack requires that the attacker has access to a user account on the TURN server to be able to set up the TURN mappings. To prevent this attack, the RTSP server needs to verify that the ultimate target destination accepts this media stream, which would require something like ICE's connectivity checks being run between the RTSP server and the RTSP client.


5. Firewalls
5. 防火墙

Firewalls exist for the purpose of protecting a network from traffic not desired by the firewall owner. Therefore, it is a policy decision if a firewall will let RTSP and its media streams through or not. RTSP is designed to be firewall friendly in that it should be easy to design firewall policies to permit passage of RTSP traffic and its media streams.


The firewall will need to allow the media streams associated with an RTSP session to pass through it. Therefore, the firewall will need an ALG that reads RTSP SETUP and TEARDOWN messages. By reading the SETUP message, the firewall can determine what type of transport and from where the media stream packets will be sent. Commonly, there will be the need to open UDP ports for RTP/RTCP. By looking at the source and destination addresses and ports, the opening in the firewall can be minimized to the least necessary. The opening in the firewall can be closed after a TEARDOWN message for that session or the session itself times out.


The above possibilities for firewalls to inspect and respond to the signaling are prevented if end-to-end confidentiality protection is used for the RTSP signaling, e.g., using the specified RTSP over TLS. As a result, firewalls can't be actively opening pinholes for the media streams based on the signaling. To enable an RTSP ALG in the firewall to correctly function, the hop-by-hop signaling security in RTSP 2.0 can be used (see Section 19.3 of [RTSP]). If not, other methods have to be used to enable the transport flows for the media.

如果对RTSP信令使用端到端保密保护(例如,通过TLS使用指定的RTSP),则阻止防火墙检查和响应信令的上述可能性。因此,防火墙无法根据信令主动为媒体流打开针孔。为了使防火墙中的RTSP ALG正常工作,可以使用RTSP 2.0中的逐跳信令安全性(参见[RTSP]第19.3节)。如果没有,则必须使用其他方法来启用介质的传输流。

Simpler firewalls do allow a client to receive media as long as it has sent packets to the target. Depending on the security level, this can have the same behavior as a NAT. The only difference is


that no address translation is done. To use such a firewall, a client would need to implement one of the above described NAT traversal methods that include sending packets to the server to create the necessary filtering state.


6. Comparison of NAT Traversal Techniques
6. NAT穿越技术的比较

This section evaluates the techniques described above against the requirements listed in Section 3.


In the following table, the columns correspond to the numbered requirements. For instance, the column under R1 corresponds to the first requirement in Section 3: must work for all flavors of NATs. The rows represent the different NAT/firewall traversal techniques. Latch is short for Latching, "V. Latch" is short for "variation of Latching" as described in Section 4.5, and "3-W Latch" is short for the Three-Way Latching described in Section 4.6.


A summary of the requirements are:


R1: Work for all flavors of NATs


R2: Must work with firewalls, including those with ALGs


R3: Should have minimal impact on clients not behind NATs, counted in minimal number of additional RTTs


R4: Should be simple to use, implement, and administer


R5: Should provide mitigation against DDoS attacks


The following considerations are also added to the requirements:


C1: Will the solution support both clients and servers behind NAT?


C2: Is the solution robust as NAT behaviors change?


               |  R1  |  R2  |  R3  |  R4  |  R5  |  C1  |  C2  |
    STUN       | No   | Yes  |  1   | Maybe| No   | No   | No   |
    Emb. STUN  | Yes  | Yes  |  2   | Maybe| No   | No   | Yes  |
    ICE        | Yes  | Yes  | 2.5  | No   | Yes  | Yes  | Yes  |
    Latch      | Yes  | Yes  |  1   | Maybe| No   | No   | Yes  |
    V. Latch   | Yes  | Yes  |  1   | Yes  | No   | No   | Yes  |
    3-W Latch  | Yes  | Yes  | 1.5  | Maybe| Yes  | No   | Yes  |
    ALG        |(Yes) | Yes  |  0   | No   | Yes  | No   | Yes  |
    TCP Tunnel | Yes  | Yes  | 1.5  | Yes  | Yes  | No   | Yes  |
    TURN       | Yes  | Yes  |  1   | No   | Yes  |(Yes) | Yes  |
               |  R1  |  R2  |  R3  |  R4  |  R5  |  C1  |  C2  |
    STUN       | No   | Yes  |  1   | Maybe| No   | No   | No   |
    Emb. STUN  | Yes  | Yes  |  2   | Maybe| No   | No   | Yes  |
    ICE        | Yes  | Yes  | 2.5  | No   | Yes  | Yes  | Yes  |
    Latch      | Yes  | Yes  |  1   | Maybe| No   | No   | Yes  |
    V. Latch   | Yes  | Yes  |  1   | Yes  | No   | No   | Yes  |
    3-W Latch  | Yes  | Yes  | 1.5  | Maybe| Yes  | No   | Yes  |
    ALG        |(Yes) | Yes  |  0   | No   | Yes  | No   | Yes  |
    TCP Tunnel | Yes  | Yes  | 1.5  | Yes  | Yes  | No   | Yes  |
    TURN       | Yes  | Yes  |  1   | No   | Yes  |(Yes) | Yes  |

Figure 1: Comparison of Fulfillment of Requirements


Looking at Figure 1, one would draw the conclusion that using TCP Tunneling or Three-Way Latching are the solutions that best fulfill the requirements. The different techniques were discussed in the MMUSIC WG. It was established that the WG would pursue an ICE-based solution due to its generality and capability of also handling servers delivering media from behind NATs. TCP Tunneling is likely to be available as an alternative, due to its specification in the main RTSP specification. Thus, it can be used if desired, and the potential downsides of using TCP is acceptable in particular deployments. When it comes to Three-Way Latching, it is a very competitive technique given that you don't need support for RTSP servers behind NATs. There was some discussion in the WG about if the increased implementation burden of ICE is sufficiently motivated compared to a the Three-Way Latching solution for this generality. In the end, the authors believed that the reuse of ICE, greater flexibility, and any way needed to deploy a new solution were the decisive factors.


The ICE-based RTSP NAT traversal solution is specified in "A Network Address Translator (NAT) Traversal mechanism for media controlled by Real-Time Streaming Protocol (RTSP)" [RTSP-NAT].

基于ICE的RTSP NAT遍历解决方案在“实时流协议(RTSP)控制的媒体的网络地址转换器(NAT)遍历机制[RTSP-NAT]中指定。

7. Security Considerations
7. 安全考虑

In the preceding sections, we have discussed security merits of the different NAT/firewall traversal methods for RTSP. In summary, the presence of NAT(s) is a security risk, as a client cannot perform source authentication of its IP address. This prevents the deployment of any future RTSP extensions providing security against the hijacking of sessions by a man in the middle.


Each of the proposed solutions has security implications. Using STUN will provide the same level of security as RTSP without transport-level security and source authentications, as long as the server does not allow media to be sent to a different IP address than the RTSP client request was sent from.


Using Latching will have a higher risk of session hijacking or DoS than normal RTSP. The reason is that there exists a probability that an attacker is able to guess the random bits that the client uses to prove its identity when creating the address bindings. This can be solved in the variation of Latching (Section 4.5) with authentication features. Still, both those variants of Latching are vulnerable against a deliberate attack from the RTSP client to redirect the media stream requested to any target assuming it can spoof the source address. This security vulnerability is solved by performing a Three-way Latching procedure as discussed in Section 4.6.


ICE resolves the binding vulnerability of Latching by using signed STUN messages, as well as requiring that both sides perform connectivity checks to verify that the target IP address in the candidate pair is both reachable and willing to respond. ICE can, however, create a significant amount of traffic if the number of candidate pairs are large. Thus, pacing is required and implementations should attempt to limit their number of candidates to reduce the number of packets.


If the signaling between the ICE peers (RTSP client and server) is not confidentiality and integrity protected, ICE is vulnerable to attacks where the candidate list is manipulated. The lack of signaling security will also simplify spoofing of STUN binding messages by revealing the secret used in signing.


The usage of an RTSP ALG does not in itself increase the risk for session hijacking. However, the deployment of ALGs as the sole mechanism for RTSP NAT traversal will prevent deployment of end-to-end encrypted RTSP signaling.

使用RTSP ALG本身不会增加会话劫持的风险。然而,将ALG部署为RTSP NAT遍历的唯一机制将阻止端到端加密RTSP信令的部署。

The usage of TCP tunneling has no known security problems. However, it might provide a bottleneck when it comes to end-to-end RTSP signaling security if TCP tunneling is used on an interleaved RTSP signaling connection.


The usage of TURN has severe risk of DoS attacks against a client. The TURN server can also be used as a redirect point in a DDoS attack unless the server has strict enough rules for who may create bindings.


Since Latching and the variants of Latching have such big security issues, they should not be used at all. Three-Way Latching as well as ICE mitigates these security issues and performs the important return-routability checks that prevent spoofed source addresses, and they should be recommended for that reason. RTP ALGs are a security risk as they can create an incitement against using secure RTSP signaling. That can be avoided as ALGs require trust in the middlebox, and that trust becomes explicit if one uses the hop-by-hop security solution as specified in Section 19.3 of RTSP 2.0. [RTSP]. The remaining methods can be considered safe enough, assuming that the appropriate security mechanisms are used and not ignored.

因为锁存和锁存的变体有如此大的安全问题,所以根本不应该使用它们。三路锁存和ICE缓解了这些安全问题,并执行重要的返回路由性检查,以防止伪造源地址,因此应建议使用这些检查。RTP ALG是一种安全风险,因为它们会煽动使用安全RTSP信号。这是可以避免的,因为ALG需要中间盒中的信任,并且如果使用RTSP 2.0第19.3节中指定的逐跳安全解决方案,则该信任将变得明确。[RTSP]。假设使用了适当的安全机制,而不是忽略,则可以认为其余的方法足够安全。

8. Informative References
8. 资料性引用

[NICE] Libnice, "The GLib ICE implementation", June 2015, <>.

[NICE]Libnice,“GLib ICE实施”,2015年6月<>.

[PJNATH] "PJNATH - Open Source ICE, STUN, and TURN Library", May 2013, <>.


[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI 10.17487/RFC0768, August 1980, <>.

[RFC768]Postel,J.,“用户数据报协议”,STD 6,RFC 768,DOI 10.17487/RFC0768,1980年8月<>.

[RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, DOI 10.17487/RFC0793, September 1981, <>.

[RFC793]Postel,J.,“传输控制协议”,标准7,RFC 793,DOI 10.17487/RFC0793,1981年9月<>.

[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, DOI 10.17487/RFC2326, April 1998, <>.

[RFC2326]Schulzrinne,H.,Rao,A.,和R.Lanphier,“实时流协议(RTSP)”,RFC 2326,DOI 10.17487/RFC2326,1998年4月<>.

[RFC2588] Finlayson, R., "IP Multicast and Firewalls", RFC 2588, DOI 10.17487/RFC2588, May 1999, <>.

[RFC2588]Finlayson,R.,“IP多播和防火墙”,RFC 2588,DOI 10.17487/RFC2588,1999年5月<>.

[RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address Translator (NAT) Terminology and Considerations", RFC 2663, DOI 10.17487/RFC2663, August 1999, <>.

[RFC2663]Srisuresh,P.和M.Holdrege,“IP网络地址转换器(NAT)术语和注意事项”,RFC 2663,DOI 10.17487/RFC2663,1999年8月<>.

[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network Address Translator (Traditional NAT)", RFC 3022, DOI 10.17487/RFC3022, January 2001, <>.

[RFC3022]Srisuresh,P.和K.Egevang,“传统IP网络地址转换器(传统NAT)”,RFC 3022,DOI 10.17487/RFC3022,2001年1月<>.

[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, June 2002, <>.

[RFC3261]Rosenberg,J.,Schulzrinne,H.,Camarillo,G.,Johnston,A.,Peterson,J.,Sparks,R.,Handley,M.,和E.Schooler,“SIP:会话启动协议”,RFC 3261,DOI 10.17487/RFC3261,2002年6月<>.

[RFC3424] Daigle, L., Ed. and IAB, "IAB Considerations for UNilateral Self-Address Fixing (UNSAF) Across Network Address Translation", RFC 3424, DOI 10.17487/RFC3424, November 2002, <>.

[RFC3424]Daigle,L.,Ed.和IAB,“网络地址转换中单边自地址固定(UNSAF)的IAB考虑”,RFC 3424DOI 10.17487/RFC3424,2002年11月<>.

[RFC3489] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy, "STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)", RFC 3489, DOI 10.17487/RFC3489, March 2003, <>.

[RFC3489]Rosenberg,J.,Weinberger,J.,Huitema,C.,和R.Mahy,“STUN-通过网络地址转换器(NAT)简单遍历用户数据报协议(UDP)”,RFC 3489,DOI 10.17487/RFC3489,2003年3月<>.

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <>.

[RFC3550]Schulzrinne,H.,Casner,S.,Frederick,R.,和V.Jacobson,“RTP:实时应用的传输协议”,STD 64,RFC 3550,DOI 10.17487/RFC3550,2003年7月<>.

[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006, <>.

[RFC4566]Handley,M.,Jacobson,V.,和C.Perkins,“SDP:会话描述协议”,RFC 4566,DOI 10.17487/RFC4566,2006年7月<>.

[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July 2006, <>.

[RFC4571]Lazzaro,J.,“面向连接传输上的帧实时传输协议(RTP)和RTP控制协议(RTCP)数据包”,RFC 4571,DOI 10.17487/RFC4571,2006年7月<>.

[RFC4787] Audet, F., Ed. and C. Jennings, "Network Address Translation (NAT) Behavioral Requirements for Unicast UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January 2007, <>.

[RFC4787]Audet,F.,Ed.和C.Jennings,“单播UDP的网络地址转换(NAT)行为要求”,BCP 127,RFC 4787,DOI 10.17487/RFC4787,2007年1月<>.

[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007, <>.

[RFC4961]Wing,D,“对称RTP/RTP控制协议(RTCP)”,BCP 131,RFC 4961,DOI 10.17487/RFC49611907年7月<>.

[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, DOI 10.17487/RFC5245, April 2010, <>.

[RFC5245]Rosenberg,J.,“交互式连接建立(ICE):提供/应答协议的网络地址转换器(NAT)遍历协议”,RFC 5245,DOI 10.17487/RFC5245,2010年4月<>.

[RFC5382] Guha, S., Ed., Biswas, K., Ford, B., Sivakumar, S., and P. Srisuresh, "NAT Behavioral Requirements for TCP", BCP 142, RFC 5382, DOI 10.17487/RFC5382, October 2008, <>.

[RFC5382]Guha,S.,Ed.,Biswas,K.,Ford,B.,Sivakumar,S.,和P.Srisuresh,“TCP的NAT行为要求”,BCP 142,RFC 5382,DOI 10.17487/RFC5382,2008年10月<>.

[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, DOI 10.17487/RFC5389, October 2008, <>.

[RFC5389]Rosenberg,J.,Mahy,R.,Matthews,P.,和D.Wing,“NAT(STUN)的会话遍历实用程序”,RFC 5389,DOI 10.17487/RFC5389,2008年10月<>.

[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)", RFC 5764, DOI 10.17487/RFC5764, May 2010, <>.

[RFC5764]McGrew,D.和E.Rescorla,“为安全实时传输协议(SRTP)建立密钥的数据报传输层安全(DTLS)扩展”,RFC 5764,DOI 10.17487/RFC5764,2010年5月<>.

[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)", RFC 5766, DOI 10.17487/RFC5766, April 2010, <>.

[RFC5766]Mahy,R.,Matthews,P.,和J.Rosenberg,“使用NAT周围的中继进行遍历(TURN):NAT(STUN)会话遍历实用程序的中继扩展”,RFC 5766,DOI 10.17487/RFC5766,2010年4月<>.

[RFC6062] Perreault, S., Ed. and J. Rosenberg, "Traversal Using Relays around NAT (TURN) Extensions for TCP Allocations", RFC 6062, DOI 10.17487/RFC6062, November 2010, <>.

[RFC6062]Perreault,S.,Ed.和J.Rosenberg,“围绕TCP分配的NAT(TURN)扩展使用中继进行遍历”,RFC 6062,DOI 10.17487/RFC6062,2010年11月<>.

[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows", RFC 6263, DOI 10.17487/RFC6263, June 2011, <>.

[RFC6263]Marjou,X.和A.Sollaud,“保持与RTP/RTP控制协议(RTCP)流相关的NAT映射活动的应用机制”,RFC 6263,DOI 10.17487/RFC6263,2011年6月<>.

[RFC6275] Perkins, C., Ed., Johnson, D., and J. Arkko, "Mobility Support in IPv6", RFC 6275, DOI 10.17487/RFC6275, July 2011, <>.

[RFC6275]Perkins,C.,Ed.,Johnson,D.,和J.Arkko,“IPv6中的移动支持”,RFC 6275,DOI 10.17487/RFC6275,2011年7月<>.

[RFC7362] Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT Traversal (HNT) for Media in Real-Time Communication", RFC 7362, DOI 10.17487/RFC7362, September 2014, <>.

[RFC7362]Ivov,E.,Kaplan,H.,和D.Wing,“闭锁:实时通信中媒体的托管NAT穿越(HNT)”,RFC 7362,DOI 10.17487/RFC7362,2014年9月<>.

[RTP-NO-OP] Andreasen, F., "A No-Op Payload Format for RTP", Work in Progress, draft-ietf-avt-rtp-no-op-04, May 2007.


[RTSP] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., and M. Stiemerling, "Real Time Streaming Protocol 2.0 (RTSP)", Work in Progress, draft-ietf-mmusic-rfc2326bis-40, February 2014.


[RTSP-NAT] Goldberg, J., Westerlund, M., and T. Zeng, "A Network Address Translator (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming Protocol (RTSP)", Work in Progress, draft-ietf-mmusic-rtsp-nat-22, July 2014.


[STUN-IMPL] "Open Source STUN Client and Server", May 2013, <>.




The authors would also like to thank all persons on the MMUSIC working group's mailing list that have commented on this document. Persons having contributed to this protocol, in no special order, are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon, Amir Wolf, Anders Klemets, Flemming Andreasen, Ari Keranen, Bill Atwood, Alissa Cooper, Colin Perkins, Sarah Banks, David Black, and Alvaro Retana. Thomas Zeng would also like to give special thanks to Greg Sherwood of PacketVideo for his input into this memo.

作者还要感谢MMUSIC工作组邮件列表中对本文件发表评论的所有人员。参与本协议的人员(无特殊顺序)为:乔纳森·罗森博格、菲利普·金特里克、汤姆·马歇尔、大卫·杨、阿米尔·沃尔夫、安德斯·克莱梅茨、弗莱明·安德烈森、阿里·凯拉宁、比尔·阿特伍德、艾莉莎·库珀、科林·珀金斯、莎拉·班克斯、大卫·布莱克和阿尔瓦罗·雷塔纳。Thomas Zeng还想特别感谢PacketVideo的Greg Sherwood为这份备忘录所做的贡献。

Section 1.1 contains text originally written for RFC 4787 by Francois Audet and Cullen Jennings.

第1.1节包含Francois Audet和Cullen Jennings最初为RFC 4787编写的文本。

Authors' Addresses


Magnus Westerlund Ericsson Farogatan 6 Stockholm SE-164 80 Sweden

Magnus Westerlund Ericsson Farogatan 6斯德哥尔摩SE-164 80瑞典

   Phone: +46 8 719 0000
   Phone: +46 8 719 0000

Thomas Zeng PacketVideo Corp