Internet Engineering Task Force (IETF)                         J-F. Mule
Request for Comments: 6271                                     CableLabs
Category: Informational                                        June 2011
ISSN: 2070-1721
Internet Engineering Task Force (IETF)                         J-F. Mule
Request for Comments: 6271                                     CableLabs
Category: Informational                                        June 2011
ISSN: 2070-1721

Requirements for SIP-Based Session Peering




This memo captures protocol requirements to enable session peering of voice, presence, instant messaging, and other types of multimedia traffic. This informational document is intended to link the various use cases described for session peering to protocol solutions.


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) 2011 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2011 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 ....................................................2
   2. Terminology .....................................................3
   3. General Requirements ............................................3
      3.1. Scope ......................................................4
      3.2. Border Elements ............................................4
      3.3. Session Establishment Data .................................8
           3.3.1. User Identities and SIP URIs ........................8
           3.3.2. URI Reachability ....................................9
   4. Requirements for Session Peering of Presence and
      Instant Messaging ..............................................10
   5. Security Considerations ........................................12
      5.1. Security Properties for the Acquisition of Session
           Establishment Data ........................................12
      5.2. Security Properties for the SIP Signaling Exchanges .......13
      5.3. End-to-End Media Security .................................14
   6. Acknowledgments ................................................15
   7. References .....................................................15
      7.1. Normative References ......................................15
      7.2. Informative References ....................................15
   Appendix A. Policy Parameters for Session Peering .................19
     A.1. Categories of Parameters for VoIP Session Peering and
          Justifications .............................................19
     A.2. Summary of Parameters for Consideration in Session
          Peering Policies ...........................................22
   1. Introduction ....................................................2
   2. Terminology .....................................................3
   3. General Requirements ............................................3
      3.1. Scope ......................................................4
      3.2. Border Elements ............................................4
      3.3. Session Establishment Data .................................8
           3.3.1. User Identities and SIP URIs ........................8
           3.3.2. URI Reachability ....................................9
   4. Requirements for Session Peering of Presence and
      Instant Messaging ..............................................10
   5. Security Considerations ........................................12
      5.1. Security Properties for the Acquisition of Session
           Establishment Data ........................................12
      5.2. Security Properties for the SIP Signaling Exchanges .......13
      5.3. End-to-End Media Security .................................14
   6. Acknowledgments ................................................15
   7. References .....................................................15
      7.1. Normative References ......................................15
      7.2. Informative References ....................................15
   Appendix A. Policy Parameters for Session Peering .................19
     A.1. Categories of Parameters for VoIP Session Peering and
          Justifications .............................................19
     A.2. Summary of Parameters for Consideration in Session
          Peering Policies ...........................................22
1. Introduction
1. 介绍

Peering at the session level represents an agreement between parties to exchange multimedia traffic. In this document, we assume that the Session Initiation Protocol (SIP) is used to establish sessions between SIP Service Providers (SSPs). SIP Service Providers are referred to as peers, and they are typically represented by users, user groups, enterprises, real-time collaboration service communities, or other service providers offering voice or multimedia services using SIP.


A number of documents have been developed to provide background information about SIP session peering. It is expected that the reader is familiar with the reference architecture described in [ARCHITECTURE], use cases for voice ([VOIP]), and instant messaging and presence ([RFC5344]).


Peering at the session layer can be achieved on a bilateral basis (direct peering established directly between two SSPs), or on an indirect basis via a session intermediary (indirect peering via a third-party SSP that has a trust relationship with the SSPs) -- see the terminology document [RFC5486] for more details.


This document first describes general requirements. The use cases are then analyzed in the spirit of extracting relevant protocol requirements that must be met to accomplish the use cases. These requirements are intended to be independent of the type of media exchanged such as Voice over IP (VoIP), video telephony, and instant messaging (IM). Requirements specific to presence and instant messaging are defined in Section 4.


It is not the goal of this document to mandate any particular use of IETF protocols other than SIP by SIP Service Providers in order to establish session peering. Instead, the document highlights what requirements should be met and what protocols might be used to define the solution space.


Finally, we conclude with a list of parameters for the definition of a session peering policy, provided in an informative appendix. It should be considered as an example of the information SIP Service Providers may have to discuss or agree on to exchange SIP traffic.


2. Terminology
2. 术语

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119].


This document also reuses the terminology defined in [RFC5486].


It is assumed that the reader is familiar with the Session Description Protocol (SDP) [RFC4566] and the Session Initiation Protocol (SIP) [RFC3261]. Finally, when used with capital letters, the term 'Authentication Service' is to be understood as defined by SIP Identity [RFC4474].


3. General Requirements
3. 一般要求

The following sub-sections contain general requirements applicable to multiple use cases for multimedia session peering.


3.1. Scope
3.1. 范围

The primary focus of this document is on the requirements applicable to the boundaries of Layer 5 SIP networks: SIP entities, signaling path border elements (SBEs), and the associated protocol requirements for the look-up and location routing of the session establishment data. The requirements applicable to SIP User Agents or related to the provisioning of the session data are considered out of scope.


SIP Service Providers have to reach an agreement on numerous points when establishing session peering relationships.


This document highlights only certain aspects of a session peering agreement. It describes the requirements relevant to protocols in four areas: the declaration, advertisement and management of ingress and egress border elements for session signaling and media (Section 3.2), the information exchange related to the Session Establishment Data (SED, Section 3.3), specific requirements for presence and instant message (Section 4), and the security properties that may be desirable to secure session exchanges (Section 5).


Numerous other considerations of session peering arrangements are critical to reach a successful agreement, but they are considered out of scope of this document. They include information about SIP protocol support (e.g., SIP extensions and field conventions), media (e.g., type of media traffic to be exchanged, compatible media codecs and transport protocols, mechanisms to ensure differentiated quality of service for media), Layer 3 IP connectivity between the signaling and data path border elements, and accounting and traffic capacity control (e.g., the maximum number of SIP sessions at each ingress point, or the maximum number of concurrent IM or VoIP sessions).


The informative Appendix A lists parameters that may be considered when discussing the technical parameters of SIP session peering. The purpose of this list is to capture the parameters that are considered outside the scope of the protocol requirements.


3.2. Border Elements
3.2. 边界元素

For border elements to be operationally manageable, maximum flexibility should be given for how they are declared or dynamically advertised. Indeed, in any session peering environment, there is a need for a SIP Service Provider to declare or dynamically advertise the SIP entities that will face the peer's network. The data path border elements are typically signaled dynamically in the session description.


The use cases defined in [VOIP] catalog the various border elements between SIP Service Providers; they include signaling path border elements (SBEs) and SIP proxies (or any SIP entity at the boundary of the Layer 5 network).


o Requirement #1:

o 要求#1:

Protocol mechanisms MUST be provided to enable a SIP Service Provider to communicate the ingress signaling path border elements of its service domain.


Notes on solution space:


The SBEs may be advertised to session peers using static mechanisms, or they may be dynamically advertised. There is general agreement that [RFC3263] provides a solution for dynamically advertising ingress SBEs in most cases of direct or indirect peering. We discuss the DNS-based solution space further in Requirement #4 below, especially in cases where the DNS response varies based on who sends the query (peer-dependent SBEs).


o Requirement #2:

o 要求#2:

Protocol mechanisms MUST be provided to enable a SIP Service Provider to communicate the egress SBEs of its service domain.


Notes on motivations for this requirement:


For the purposes of capacity planning, traffic engineering, and call admission control, a SIP Service Provider may be asked from where it will generate SIP calls. The SSP accepting calls from a peer may wish to know from where SIP calls will originate (this information is typically used by the terminating SSP).


While provisioning requirements are out of scope, some SSPs may find use for a mechanism to dynamically advertise or discover the egress SBEs of a peer.


If the SSP also provides media streams to its users as shown in the use cases for "originating" and "terminating" SSPs, a mechanism must exist to allow SSPs to advertise their egress and ingress data path border elements (DBEs), if applicable. While some SSPs may have open policies and accept media traffic from anywhere outside their network to anywhere inside their network, some SSPs may want to optimize media delivery and identify media paths between peers prior to traffic being sent (Layer 5 to Layer 3 Quality of Service (QoS) mapping).


o Requirement #3:

o 要求#3:

Protocol mechanisms MUST be provided to allow a SIP Service Provider to communicate its DBEs to its peers.


Notes: Some SSPs engaged in SIP interconnects do exchange this type of DBE information in a static manner. Some SSPs do not.


In some SIP networks, SSPs may expose the same border elements to all peers. In other environments, it is common for SSPs to advertise specific SBEs and DBEs to certain peers. This is done by SSPs to meet specific objectives for a given peer: routing optimization of the signaling and media exchanges, optimization of the latency or throughput based on the 'best' SBE and DBE combination, and other service provider policy parameters. These are some of the reasons why advertisement of SBEs and DBEs may be peer dependent.


o Requirement #4:

o 要求#4:

The mechanisms recommended for the declaration or advertisement of SBE and DBE entities MUST allow for peer variability.


Notes on solution space:


A simple solution is to advertise SBE entities using DNS and [RFC3263] by providing different DNS names to different peers. This approach has some practical limitations because the SIP URIs containing the DNS names used to resolve the SBEs may be propagated by users, for example, in the form of sip:user@domain. It is impractical to ask users to implement different target URIs based upon their SIP Service Provider's desire to receive incoming session signaling at different ingress SBEs based upon the originator. The solution described in [RFC3263] and based on DNS to advertise SBEs is therefore under specified for this requirement.

一个简单的解决方案是通过向不同的对等方提供不同的DNS名称,使用DNS和[RFC3263]公布SBE实体。此方法具有一些实际限制,因为包含用于解析SBE的DNS名称的SIP URI可能由用户传播,例如,以SIP的形式:user@domain. 要求用户根据其SIP服务提供商的愿望来实现不同的目标URI是不切实际的,因为他们的SIP服务提供商希望在不同的入口SBE(基于发起者)接收传入的会话信令。因此,[RFC3263]中描述的基于DNS的SBE广告解决方案不符合此要求。

Other DNS mechanisms have been used extensively in other areas of the Internet, in particular in Content Distribution Internetworking to make the DNS responses vary based on the originator of the DNS query (see [RFC3466], [RFC3568], and [RFC3570]). The applicability of such solutions for session peering needs further analysis.


Finally, other techniques such as Anycast services ([RFC4786]) may be employed at lower layers than Layer 5 to provide a solution to this requirement. For example, anycast nodes could be defined by SIP service providers to expose a common address for SBEs into DNS, allowing the resolution of the anycast node address to the


appropriate peer-dependent service address based on the routing topology or other criteria gathered from the combined use of anycast and DNS techniques.


Notes on variability of the SBE advertisements based on the media capabilities:


Some SSPs may have some restrictions on the type of media traffic their SBEs can accept. For SIP sessions however, it is not possible to communicate those restrictions in advance of the session initiation: a SIP target may support voice-only media, voice and video, or voice and instant messaging communications. While the inability to find out whether a particular type of SIP session can be terminated by a certain SBE can cause session attempts to fail, there is consensus to not add a new requirement in this document. These aspects are essentially covered by SSPs when discussing traffic exchange policies and are deemed out of scope of this document.


In the use cases provided as part of direct and indirect peering scenarios, an SSP deals with multiple SIP entities and multiple SBEs in its own domain. There is often a many-to-many relationship between the SIP proxies considered inside the trusted network boundary of the SSP and its signaling path border elements at the network boundaries.


It should be possible for an SSP to define which egress SBE a SIP entity must use based on a given peer destination.


For example, in the case of a static direct peering scenario (Figure 2 in Section 5.2. of [VOIP]), it should be possible for the SIP proxy in the originating network (O-Proxy) to select the appropriate egress SBE (O-SBE) to reach the SIP target based on the information the proxy receives from the Look-Up Function (O-LUF), and/or Location Routing Function (O-LRF) -- message response labeled (2). Note that this example also applies to the case of indirect peering when a service provider has multiple service areas and each service area involves multiple SIP proxies and a few SBEs.


o Requirement #5:

o 要求#5:

The mechanisms recommended for the Look-Up Function (LUF) and the Location Routing Functions (LRF) MUST be capable of returning both a target URI destination and a value providing the next SIP hop(s).


Notes: solutions may exist depending on the choice of the protocol used between the Proxy and its LUF/LRF. The idea is for the O-Proxy to be provided with the next SIP hop and the equivalent of one or more SIP Route header values. If ENUM is used as a protocol for the LUF, the solution space is undefined.


It is desirable for an SSP to be able to communicate how authentication of a peer's SBEs will occur (see the security requirements for more details).


o Requirement #6:

o 要求#6:

The mechanisms recommended for locating a peer's SBE MUST be able to convey how a peer should initiate secure session establishment.


Notes: some mechanisms exist. For example, the required use of SIP over TLS may be discovered via [RFC3263], and guidelines concerning the use of the SIPS URI scheme in SIP have been documented in [RFC5630].

注:存在一些机制。例如,通过[RFC3263]可以发现需要在TLS上使用SIP,关于在SIP中使用SIPS URI方案的指南已记录在[RFC5630]中。

3.3. Session Establishment Data
3.3. 会话建立数据

The Session Establishment Data (SED) is defined in [RFC5486] as the data used to route a call to the next hop associated with the called domain's ingress point. The following paragraphs capture some general requirements on the SED data.


3.3.1. User Identities and SIP URIs
3.3.1. 用户身份和SIPURI

User identities used between peers can be represented in many different formats. Session Establishment Data should rely on URIs (Uniform Resource Identifiers, [RFC3986]) and SIP URIs should be preferred over tel URIs ([RFC3966]) for session peering of VoIP traffic.

对等方之间使用的用户身份可以用许多不同的格式表示。会话建立数据应依赖于URI(统一资源标识符[RFC3986]),对于VoIP流量的会话对等,SIP URI应优先于tel URI([RFC3966])。

The use of DNS domain names and hostnames is recommended in SIP URIs and they should be resolvable on the public Internet. As for the user part of the SIP URIs, the mechanisms for session peering should not require an SSP to be aware of which individual user identities are valid within its peer's domain.

建议在SIP URI中使用DNS域名和主机名,并且它们应该可以在公共Internet上解析。对于SIP uri的用户部分,会话对等机制不应要求SSP知道在其对等方的域中哪些用户身份是有效的。

o Requirement #7:

o 要求#7:

The protocols used for session peering MUST accommodate the use of different types of URIs. URIs with the same domain-part SHOULD share the same set of peering policies; thus, the domain of the SIP URI may be used as the primary key to any information

用于会话对等的协议必须适应不同类型URI的使用。具有相同域部分的URI应共享相同的对等策略集;因此,SIP URI的域可以用作任何信息的主键

regarding the reachability of that SIP URI. The host part of SIP URIs SHOULD contain a fully qualified domain name instead of a numeric IPv4 or IPv6 address.

关于SIPURI的可达性。SIP URI的主机部分应包含完全限定的域名,而不是数字IPv4或IPv6地址。

o Requirement #8:

o 要求#8:

The mechanisms for session peering should not require an SSP to be aware of which individual user identities are valid within its peer's domain.


o Notes on the solution space for Requirements #7 and #8:

o 关于需求#7和#8的解决方案空间的注释:

This is generally well supported by IETF protocols. When telephone numbers are in tel URIs, SIP requests cannot be routed in accordance with the traditional DNS resolution procedures standardized for SIP as indicated in [RFC3824]. This means that the solutions built for session peering must not solely use Public Switched Telephone Network (PSTN) identifiers such as Service Provider IDs (SPIDs) or Trunk Group IDs (they should not be precluded but solutions should not be limited to these).




Although SED data may be based on E.164-based SIP URIs for voice interconnects, a generic peering methodology should not rely on such E.164 numbers.

尽管SED数据可能基于用于语音互连的基于E.164的SIP URI,但通用对等方法不应依赖于此类E.164号码。

3.3.2. URI Reachability
3.3.2. URI可达性

Based on a well-known URI type (e.g., sip:, pres:, or im: URIs), it must be possible to determine whether the SSP domain servicing the URI allows for session peering, and if it does, it should be possible to locate and retrieve the domain's policy and SBE entities.


For example, an originating service provider must be able to determine whether a SIP URI is open for direct interconnection without requiring an SBE to initiate a SIP request. Furthermore, since each call setup implies the execution of any proposed algorithm, the establishment of a SIP session via peering should incur minimal overhead and delay, and employ caching wherever possible to avoid extra protocol round trips.

例如,发起服务提供商必须能够确定SIP URI是否为直接互连而开放,而无需SBE发起SIP请求。此外,由于每个呼叫设置都意味着执行任何提出的算法,因此通过对等建立SIP会话应产生最小的开销和延迟,并尽可能使用缓存来避免额外的协议往返。

o Requirement #9:

o 要求#9:

The mechanisms for session peering MUST allow an SBE to locate its peer SBE given a URI type and the target SSP domain name.


4. Requirements for Session Peering of Presence and Instant Messaging
4. 状态和即时消息会话对等的要求

This section describes requirements for presence and instant messaging session peering.


Two SSPs create a peering relationship to enable their IM and presence users to collaborate with users on the other SSP network. We focus the requirements on inter-domain subscriptions to presence information, the exchange of messages and privacy settings, and the use of standard presence document formats across domains.


Several use cases for presence and instant messaging peering are described in [RFC5344], a document authored by A. Houri, E. Aoki, and S. Parameswar. Credits for the original content captured from these use cases into requirements in this section must go to them.


o Requirement #10:

o 要求#10:

The mechanisms recommended for the exchange of presence information between SSPs SHOULD allow a user of one presence community to send a presence subscription request to presentities served by another SSP via its local community, including subscriptions to a single presentity, a personal, public or ad hoc group list of presentities.


Notes: see Sections 2.1 and 2.2 of [RFC5344].


o Requirement #11:

o 要求#11:

The mechanisms recommended for instant messaging exchanges between SSPs SHOULD allow a user of one SSP's community to communicate with users of the other SSP community via their local community using the various methods. Note that some SSPs may exercise some control over which methods are allowed based on service policies. Such methods include sending a one-time IM message, initiating a SIP session for transporting sessions of messages, participating in n-way chats using chat rooms with users from the peer SSPs, etc.


Notes: see Sections 2.4, 2.5, and 2.6 of [RFC5344].


o Requirement #12:

o 要求#12:

In some presence communities, users can define the list of watchers that receive presence notifications for a given presentity. Such privacy settings for watcher notifications per presentity are typically not shared across SSPs causing multiple notifications to be sent for one presentity change between SSPs.


The sharing of those privacy settings per presentity between SSPs would allow fewer notifications: a single notification would be sent per presentity and the terminating SSP would send notifications to the appropriate watchers according to the presentity's privacy information.


The mechanisms recommended for presence information exchanges between SSPs SHOULD allow the sharing of some user privacy settings in order for users to convey the list of watchers that can receive notification of presence information changes on a per-presentity basis.


The privacy sharing mechanism must be done with the express consent of the user whose privacy settings will be shared with the other community. Because of the privacy-sensitive information exchanged between SSPs, the protocols used for the exchange of presence information must follow the security recommendations defined in Section 6 of [RFC3863].


Notes: see Section 2.3 of [RFC5344].


o Requirement #13:

o 要求#13:

It should be possible for an SSP to associate a presence document with a list of watchers in the peer SSP community so that the peer watchers can receive the presence document notifications. This will enable sending less presence document notifications between the communities while avoiding the need to share privacy information of presentities from one community to the other.


The systems used to exchange presence documents between SSPs SHOULD allow a presence document to be delivered to one or more watchers.


Note: The presence document and the list of authorized watchers in the peer SSP may be sent separately. Also, the privacy-sharing mechanisms defined in Requirement #12 also apply to this requirement.


o Requirement #14:

o 要求#14:

Early deployments of SIP-based presence and instant messaging gateways have been done in front of legacy proprietary systems that use different naming schemes or name values for the elements and properties defined in a Presence Information Data Format (PIDF) document ([RFC3863]). For example, the value "Do Not Disturb" in one presence service may be mapped to "Busy" in


another system for the status element. Beyond this example of status values, it is important to ensure that the meaning of the presence information is preserved between SSPs.


The systems used to exchange presence documents between SSPs SHOULD use standard PIDF documents and translate any non-standard value of a PIDF element to a standard one.


5. Security Considerations
5. 安全考虑

This section describes the security properties that are desirable for the protocol exchanges in scope of session peering. Three types of information flows are described in the architecture and use case documents: the acquisition of the Session Establishment Data (SED) based on a destination target via the Look-Up and Location Routing Functions (LUF and LRF), the SIP signaling between SIP Service Providers, and the associated media exchanges.


This section is focused on three security services: authentication, data confidentiality, and data integrity as summarized in [RFC3365]. However, this text does not specify the mandatory-to-implement security mechanisms as required by [RFC3365]; this is left for future protocol solutions that meet the requirements.


A security threat analysis provides additional guidance for session peering ([VOIPTHREATS]).


5.1. Security Properties for the Acquisition of Session Establishment Data

5.1. 获取会话建立数据的安全属性

The Look-Up Function (LUF) and Location Routing Function (LRF) are defined in [RFC5486]. They provide mechanisms for determining the SIP target address and domain the request should be sent to, and the associated SED to route the request to that domain.


o Requirement #15:

o 要求#15:

The protocols used to query the Look-Up and Location Routing Functions SHOULD support mutual authentication.




A mutual authentication service should be provided for the LUF and LRF protocol exchanges. The content of the response returned by the LUF and LRF may depend on the identity of the requestor: the authentication of the LUF and LRF requests is therefore a desirable property. Mutual authentication is also desirable: the requestor may verify the identity of the systems that provided the


LUF and LRF responses given the nature of the data returned in those responses. Authentication also provides some protection for the availability of the LUF and LRF against attackers that would attempt to launch Denial-of-Service (DoS) attacks by sending bogus requests causing the LUF to perform a lookup and consume resources.


o Requirement #16:

o 要求#16:

The protocols used to query the Look-Up and Location Routing Functions SHOULD provide support for data confidentiality and integrity.




Given the sensitive nature of the session establishment data exchanged with the LUF and LRF functions, the protocol mechanisms chosen for the look-up and location routing should offer data confidentiality and integrity protection (SED data may contain user addresses, SIP URI, location of SIP entities at the boundaries of SIP Service Provider domains, etc.).

鉴于与LUF和LRF功能交换的会话建立数据的敏感性,为查找和位置路由选择的协议机制应提供数据机密性和完整性保护(SED数据可能包含用户地址、SIP URI、SIP实体在SIP服务提供商域边界上的位置等)。

o Notes on the solution space for Requirements #15 and #16:

o 关于需求#15和#16的解决方案空间的注释:

ENUM, SIP, and proprietary protocols are typically used today for accessing these functions. Even though SSPs may use lower-layer security mechanisms to guarantee some of those security properties, candidate protocols for the LUF and LRF should meet the above requirements.


5.2. Security Properties for the SIP Signaling Exchanges
5.2. SIP信令交换的安全属性

The SIP signaling exchanges are out of scope of this document. This section describes some of the security properties that are desirable in the context of SIP interconnects between SSPs without formulating any normative requirements.


In general, the security properties desirable for the SIP exchanges in an inter-domain context apply to session peering. These include:


o securing the transport of SIP messages between the peers' SBEs. Authentication of SIP communications is desirable, especially in the context of session peering involving SIP intermediaries. Data confidentiality and integrity of the SIP message body may be desirable as well given some of the levels of session peering indirection (indirect/assisted peering), but they could be harmful as they may prevent intermediary SSPs from "inserting" SBEs/DBEs along the signaling and data paths.

o 保护对等方SBE之间SIP消息的传输。SIP通信的认证是可取的,尤其是在涉及SIP中介的会话对等的上下文中。考虑到会话间接对等(间接/辅助对等)的一些级别,SIP消息体的数据机密性和完整性也可能是可取的,但是它们可能是有害的,因为它们可能阻止中间ssp沿信令和数据路径“插入”sbe/dbe。

o providing an Authentication Service to authenticate the identity of connected users based on the SIP Service Provider domains (for both the SIP requests and the responses).

o 提供身份验证服务,以基于SIP服务提供商域(针对SIP请求和响应)对连接用户的身份进行身份验证。

The fundamental mechanisms for securing SIP between proxy servers intra- and inter-domain are applicable to session peering; refer to Section 26.2 of [RFC3261] for transport-layer security of SIP messages using TLS, [RFC5923] for establishing TLS connections between proxies, [RFC4474] for the protocol mechanisms to verify the identity of the senders of SIP requests in an inter-domain context, and [RFC4916] for verifying the identity of the sender of SIP responses).


5.3. End-to-End Media Security
5.3. 端到端媒体安全

Media security is critical to guarantee end-to-end confidentiality of the communication between the end-users' devices, independently of how many direct or indirect peers are present along the signaling path. A number of desirable security properties emerge from this goal.


The establishment of media security may be achieved along the media path and not over the signaling path given the indirect peering use cases.


For example, media carried over the Real-Time Protocol (RTP) can be secured using secure RTP (SRTP [RFC3711]). A framework for establishing SRTP security using Datagram TLS (DTLS) [RFC4347] is described in [RFC5763]: it allows for end-to-end media security establishment using extensions to DTLS ([RFC5764]).


It should also be noted that media can be carried in numerous protocols other than RTP such as SIP (SIP MESSAGE method), MSRP (the Message Session Relay Protocol, [RFC4975], XMPP (the Extensible Messaging and Presence Protocol, [RFC6120]), and many others. Media may also be carried over TCP ([RFC4571]), and it can be encrypted over secure connection-oriented transport sessions over TLS ([RFC4572]).


A desirable security property for session peering is for SIP entities to be transparent to the end-to-end media security negotiations: SIP entities should not intervene in the Session Description Protocol (SDP) exchanges for end-to-end media security.


o Requirement #17:

o 要求#17:

The protocols used to enable session peering MUST NOT interfere with the exchanges of media security attributes in SDP. Media attribute lines that are not understood by SBEs MUST be ignored and passed along the signaling path untouched.


6. Acknowledgments
6. 致谢

This document is based on the input and contributions made by a large number of people including: Bernard Aboba, Edwin Aoki, Scott Brim, John Elwell, Patrik Faltstrom, Mike Hammer, Avshalom Houri, Otmar Lendl, Jason Livingood, Daryl Malas, Dave Meyer, Bob Natale, Sriram Parameswar, Jon Peterson, Benny Rodrig, Brian Rosen, Eric Rosenfeld, Peter Saint-Andre, David Schwartz, Richard Shocky, Henry Sinnreich, Richard Stastny, and Adam Uzelac.


Specials thanks go to Rohan Mahy, Brian Rosen, and John Elwell for their initial documents describing guidelines or best current practices in various environments, to Avshalom Houri, Edwin Aoki, and Sriram Parameswar for authoring the presence and instant messaging requirements, and to Dan Wing for providing detailed feedback on the Security Consideration sections.

特别感谢Rohan Mahy、Brian Rosen和John Elwell提供了描述各种环境中的指导原则或最佳实践的初始文档,感谢Avshalom Houri、Edwin Aoki和Sriram Parameswar编写了状态和即时消息要求,并向Dan Wing提供关于安全考虑部分的详细反馈。

7. References
7. 工具书类
7.1. Normative References
7.1. 规范性引用文件

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.

[RFC2119]Bradner,S.,“RFC中用于表示需求水平的关键词”,BCP 14,RFC 2119,1997年3月。

7.2. Informative References
7.2. 资料性引用

[ARCHITECTURE] Malas, D. and J. Livingood, "Session PEERing for Multimedia INTerconnect Architecture", Work in Progress, February 2011.


[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, September 1997.

[RFC2198]Perkins,C.,Kouvelas,I.,Hodson,O.,Hardman,V.,Handley,M.,Bolot,J.,Vega Garcia,A.,和S.Fosse Parisis,“冗余音频数据的RTP有效载荷”,RFC 21981997年9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, 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 3261,2002年6月。

[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol (SIP): Locating SIP Servers", RFC 3263, June 2002.

[RFC3263]Rosenberg,J.和H.Schulzrinne,“会话启动协议(SIP):定位SIP服务器”,RFC 3263,2002年6月。

[RFC3365] Schiller, J., "Strong Security Requirements for Internet Engineering Task Force Standard Protocols", BCP 61, RFC 3365, August 2002.

[RFC3365]Schiller,J.“互联网工程任务组标准协议的强大安全要求”,BCP 61,RFC 3365,2002年8月。

[RFC3455] Garcia-Martin, M., Henrikson, E., and D. Mills, "Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)", RFC 3455, January 2003.

[RFC3455]Garcia Martin,M.,Henrikson,E.,和D.Mills,“第三代合作伙伴关系项目(3GPP)会话启动协议(SIP)的专用头(P头)扩展”,RFC 3455,2003年1月。

[RFC3466] Day, M., Cain, B., Tomlinson, G., and P. Rzewski, "A Model for Content Internetworking (CDI)", RFC 3466, February 2003.

[RFC3466]Day,M.,Cain,B.,Tomlinson,G.,和P.Rzewski,“内容互联网(CDI)模型”,RFC 3466,2003年2月。

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, 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 35502003年7月。

[RFC3568] Barbir, A., Cain, B., Nair, R., and O. Spatscheck, "Known Content Network (CN) Request-Routing Mechanisms", RFC 3568, July 2003.

[RFC3568]Barbir,A.,Cain,B.,Nair,R.,和O.Spatscheck,“已知内容网络(CN)请求路由机制”,RFC 3568,2003年7月。

[RFC3570] Rzewski, P., Day, M., and D. Gilletti, "Content Internetworking (CDI) Scenarios", RFC 3570, July 2003.

[RFC3570]Rzewski,P.,Day,M.,和D.Gilletti,“内容互联网(CDI)场景”,RFC 35702003年7月。

[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003.

[RFC3611]Friedman,T.,Caceres,R.,和A.Clark,“RTP控制协议扩展报告(RTCP XR)”,RFC 36112003年11月。

[RFC3702] Loughney, J. and G. Camarillo, "Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP)", RFC 3702, February 2004.

[RFC3702]Loughney,J.和G.Camarillo,“会话启动协议(SIP)的身份验证、授权和记帐要求”,RFC 3702,2004年2月。

[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004.

[RFC3711]Baugher,M.,McGrew,D.,Naslund,M.,Carrara,E.,和K.Norrman,“安全实时传输协议(SRTP)”,RFC 37112004年3月。

[RFC3824] Peterson, J., Liu, H., Yu, J., and B. Campbell, "Using E.164 numbers with the Session Initiation Protocol (SIP)", RFC 3824, June 2004.

[RFC3824]Peterson,J.,Liu,H.,Yu,J.,和B.Campbell,“在会话启动协议(SIP)中使用E.164号码”,RFC 38242004年6月。

[RFC3863] Sugano, H., Fujimoto, S., Klyne, G., Bateman, A., Carr, W., and J. Peterson, "Presence Information Data Format (PIDF)", RFC 3863, August 2004.

[RFC3863]Sugano,H.,Fujimoto,S.,Klyne,G.,Batman,A.,Carr,W.,和J.Peterson,“状态信息数据格式(PIDF)”,RFC 38632004年8月。

[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC 3966, December 2004.

[RFC3966]Schulzrinne,H.,“电话号码的电话URI”,RFC 3966,2004年12月。

[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986, January 2005.

[RFC3986]Berners Lee,T.,Fielding,R.,和L.Masinter,“统一资源标识符(URI):通用语法”,STD 66,RFC 3986,2005年1月。

[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, April 2006.

[RFC4347]Rescorla,E.和N.Modadugu,“数据报传输层安全”,RFC 4347,2006年4月。

[RFC4474] Peterson, J. and C. Jennings, "Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)", RFC 4474, August 2006.

[RFC4474]Peterson,J.和C.Jennings,“会话启动协议(SIP)中身份验证管理的增强”,RFC 4474,2006年8月。

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


[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport", RFC 4571, 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 4571,2006年7月。

[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)", RFC 4572, July 2006.

[RFC4572]Lennox,J.,“会话描述协议(SDP)中传输层安全(TLS)协议上的面向连接的媒体传输”,RFC 4572,2006年7月。

[RFC4786] Abley, J. and K. Lindqvist, "Operation of Anycast Services", BCP 126, RFC 4786, December 2006.

[RFC4786]Abley,J.和K.Lindqvist,“任意广播服务的运营”,BCP 126,RFC 4786,2006年12月。

[RFC4916] Elwell, J., "Connected Identity in the Session Initiation Protocol (SIP)", RFC 4916, June 2007.

[RFC4916]Elwell,J.,“会话启动协议(SIP)中的连接身份”,RFC 4916,2007年6月。

[RFC4975] Campbell, B., Mahy, R., and C. Jennings, "The Message Session Relay Protocol (MSRP)", RFC 4975, September 2007.

[RFC4975]Campbell,B.,Mahy,R.,和C.Jennings,“消息会话中继协议(MSRP)”,RFC 49752007年9月。

[RFC5344] Houri, A., Aoki, E., and S. Parameswar, "Presence and Instant Messaging Peering Use Cases", RFC 5344, October 2008.

[RFC5344]Houri,A.,Aoki,E.,和S.Parameswar,“状态和即时消息对等使用案例”,RFC 5344,2008年10月。

[RFC5411] Rosenberg, J., "A Hitchhiker's Guide to the Session Initiation Protocol (SIP)", RFC 5411, February 2009.

[RFC5411]Rosenberg,J.,“会话启动协议(SIP)搭便车指南”,RFC 5411,2009年2月。

[RFC5486] Malas, D. and D. Meyer, "Session Peering for Multimedia Interconnect (SPEERMINT) Terminology", RFC 5486, March 2009.

[RFC5486]Malas,D.和D.Meyer,“多媒体互连的会话对等(SPEERMINT)术语”,RFC 54862009年3月。

[RFC5503] Andreasen, F., McKibben, B., and B. Marshall, "Private Session Initiation Protocol (SIP) Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture", RFC 5503, March 2009.

[RFC5503]Andreasen,F.,McKibben,B.,和B.Marshall,“支持分组电缆分布式呼叫信令体系结构的专用会话发起协议(SIP)代理到代理扩展”,RFC 5503,2009年3月。

[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)", RFC 5630, October 2009.

[RFC5630]Audet,F.“会话启动协议(SIP)中SIPS URI方案的使用”,RFC 5630,2009年10月。

[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)", RFC 5763, May 2010.

[RFC5763]Fischl,J.,Tschofenig,H.,和E.Rescorla,“使用数据报传输层安全性(DTLS)建立安全实时传输协议(SRTP)安全上下文的框架”,RFC 5763,2010年5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, 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 5764,2010年5月。

[RFC5923] Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse in the Session Initiation Protocol (SIP)", RFC 5923, June 2010.

[RFC5923]Gurbani,V.,Mahy,R.,和B.Tate,“会话启动协议(SIP)中的连接重用”,RFC 59232010年6月。

[RFC6076] Malas, D. and A. Morton, "Basic Telephony SIP End-to-End Performance Metrics", RFC 6076, January 2011.

[RFC6076]Malas,D.和A.Morton,“基本电话SIP端到端性能指标”,RFC 60762011年1月。

[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence Protocol (XMPP): Core", RFC 6120, March 2011.

[RFC6120]Saint Andre,P.,“可扩展消息和状态协议(XMPP):核心”,RFC61202011年3月。

[VOIP] Uzelac, A. and Y. Lee, "VoIP SIP Peering Use Cases", Work in Progress, April 2010.

[VOIP]Uzelac,A.和Y.Lee,“VOIP SIP对等用例”,正在进行的工作,2010年4月。

[VOIPTHREATS] Seedorf, J., Niccolini, S., Chen, E., and H. Scholz, "Session Peering for Multimedia Interconnect (SPEERMINT) Security Threats and Suggested Countermeasures", Work in Progress, March 2011.


Appendix A. Policy Parameters for Session Peering

This informative appendix lists various types of parameters that should be considered by implementers when deciding what configuration variables to expose to system administrators or management stations, as well as SSPs or federations of SSPs when discussing the technical part of a session peering policy.


In the context of session peering, a policy can be defined as the set of parameters and other information needed by an SSP to exchange traffic with another peer. Some of the session policy parameters may be statically exchanged and set throughout the lifetime of the peering relationship. Other parameters may be discovered and updated dynamically using some explicit protocol mechanisms. These dynamic parameters may be session dependent, or they may apply over multiple sessions or peers.


Various types of policy information may need to be discovered or exchanged in order to establish session peering. At a minimum, a policy should specify information related to session establishment data in order to avoid session establishment failures. A policy may also include information related to QoS, billing and accounting, and Layer 3 related interconnect requirements, which are out of the scope of this document.


Some aspects of session peering policies must be agreed to and manually implemented; they are static and are typically documented as part of a business contract, technical document, or agreement between parties. For some parameters linked to protocol support and capabilities, standard ways of expressing those policy parameters may be defined among SSPs and exchanged dynamically. For example, templates could be created in various document formats so that it could be possible to dynamically discover some of the domain policy. Such templates could be initiated by implementers. For each software or hardware release, the template could list supported RFCs, and the associated RFC parameters implemented in the given release in a standard format. Each SSP would then complete the template and adapt its content based on its service description, the deployed server or device configurations and the variation of these configurations based on peer relationships.


A.1. Categories of Parameters for VoIP Session Peering and Justifications

A.1. VoIP会话对等的参数类别和理由

The following list should be considered as an initial list of "discussion topics" to be addressed by peers when initiating a VoIP peering relationship.


o IP Network Connectivity:

o IP网络连接:

Session peers should define the IP network connectivity between their respective SBEs and DBEs. While this is out of scope of session peering, SSPs must agree on a common mechanism for IP transport of session signaling and media. This may be accomplished via private (e.g., IPVPN, IPsec, etc.) or public IP networks.


o Media-related Parameters:

o 媒体相关参数:

* Media Codecs: list of supported media codecs for audio, real-time fax (version of T.38, if applicable), real-time text (RFC 4103), dual-tone multi-frequency (DTMF) transport voice band data communications (as applicable) along with the supported or recommended codec packetization rates, level of RTP payload redundancy, audio volume levels, etc.

* 媒体编解码器:支持的音频、实时传真(T.38版本,如适用)、实时文本(RFC 4103)、双音多频(DTMF)传输声带数据通信(如适用)的媒体编解码器列表,以及支持的或建议的编解码器分组率、RTP负载冗余级别、音频音量级别等。

* Media Transport: level of support for RTP-RTCP [RFC3550], RTP Redundancy (RTP Payload for Redundant Audio Data [RFC2198]), T.38 transport over RTP, etc.

* 媒体传输:对RTP-RTCP[RFC3550]的支持级别、RTP冗余(冗余音频数据的RTP有效载荷[RFC2198])、RTP上的T.38传输等。

* Media variability at the signaling path border elements: list of media types supported by the various ingress points of a peer's network.

* 信令路径边界元素处的媒体可变性:对等网络的各种入口点支持的媒体类型列表。

* Other: support of the VoIP metric block as defined in RTP Control Protocol Extended Reports [RFC3611], etc.

* 其他:支持RTP控制协议扩展报告[RFC3611]中定义的VoIP度量块,等等。

o SIP:

o 抿:

* A session peering policy should include the list of supported and required SIP RFCs, supported and required SIP methods (including private p headers if applicable), error response codes, supported or recommended format of some header field values, etc.

* 会话对等策略应包括支持和要求的SIP RFC列表、支持和要求的SIP方法(包括专用p报头,如果适用)、错误响应代码、某些报头字段值的支持或建议格式等。

* It should also be possible to describe the list of supported SIP RFCs by various functional groupings. A group of SIP RFCs may represent how a call feature is implemented (call hold, transfer, conferencing, etc.), or it may indicate a functional grouping as in [RFC5411].

* 还可以按各种功能分组描述受支持的SIP RFC列表。SIP RFC组可以表示如何实现呼叫功能(呼叫保持、转接、会议等),也可以表示[RFC5411]中所述的功能分组。

o Accounting:

o 会计:

Methods used for call or session accounting should be specified. An SSP may require a peer to track session usage. It is critical for peers to determine whether the support of any SIP extensions for accounting is a pre-requisite for SIP interoperability. In some cases, call accounting may feed data for billing purposes, but not always: some operators may decide to use accounting as a 'bill and keep' model to track session usage and monitor usage against service level agreements.


[RFC3702] defines the terminology and basic requirements for accounting of SIP sessions. A few private SIP extensions have also been defined and used over the years to enable call accounting between SSP domains such as the P-Charging* headers in [RFC3455], the P-DCS-Billing-Info header in [RFC5503], etc.


o Performance Metrics:

o 绩效指标:

Layer 5 performance metrics should be defined and shared between peers. The performance metrics apply directly to signaling or media; they may be used proactively to help avoid congestion, call quality issues, or call signaling failures, and as part of monitoring techniques, they can be used to evaluate the performance of peering exchanges.


Examples of SIP performance metrics include the maximum number of SIP transactions per second on per-domain basis, Session Completion Rate (SCR), Session Establishment Rate (SER), etc. Some SIP end-to-end performance metrics are defined in [RFC6076]; a subset of these may be applicable to session peering and interconnects.


Some media-related metrics for monitoring VoIP calls have been defined in the VoIP Metrics Report Block, in Section 4.7 of [RFC3611].


o Security:

o 安全:

An SSP should describe the security requirements that other peers must meet in order to terminate calls to its network. While such a list of security-related policy parameters often depends on the security models pre-agreed to by peers, it is expected that these parameters will be discoverable or signaled in the future to allow session peering outside SSP clubs. The list of security parameters may be long and composed of high-level requirements (e.g., authentication, privacy, secure transport) and low-level protocol configuration elements like TLS parameters.


The following list is not intended to be complete, it provides a preliminary list in the form of examples:


* Call admission requirements: for some providers, sessions can only be admitted if certain criteria are met. For example, for some providers' networks, only incoming SIP sessions signaled over established IPsec tunnels or presented to the well-known TLS ports are admitted. Other call admission requirements may be related to some performance metrics as described above.

* 呼叫接纳要求:对于某些提供商,只有在满足某些标准的情况下才能接纳会话。例如,对于某些提供商的网络,仅允许通过已建立的IPsec隧道发送信号或呈现给已知TLS端口的传入SIP会话。其他呼叫接纳要求可能与上述一些性能指标有关。

Finally, it is possible that some requirements be imposed on lower layers, but these are considered out of scope of session peering.


* Call authorization requirements and validation: the presence of a caller or user identity may be required by an SSP. Indeed, some SSPs may further authorize an incoming session request by validating the caller's identity against white/black lists maintained by the service provider or users (traditional caller ID screening applications or IM white lists).

* 呼叫授权要求和验证:SSP可能要求有呼叫者或用户身份。实际上,一些ssp可以通过对照服务提供商或用户维护的白/黑名单(传统的呼叫者ID屏蔽应用程序或IM白名单)验证呼叫者的身份来进一步授权传入会话请求。

* Privacy requirements: an SSP may demand that its SIP messages be securely transported by its peers for privacy reasons so that the calling/called party information be protected. Media sessions may also require privacy, and some SSP policies may include requirements on the use of secure media transport protocols such as SRTP, along with some constraints on the minimum authentication/encryption options for use in SRTP.

* 隐私要求:SSP可能出于隐私原因要求其对等方安全传输其SIP消息,以便保护主叫方/被叫方信息。媒体会话也可能需要隐私,一些SSP策略可能包括对使用安全媒体传输协议(如SRTP)的要求,以及对SRTP中使用的最低身份验证/加密选项的一些限制。

* Network-layer security parameters: this covers how IPsec security associations may be established, the IPsec key exchange mechanisms should be used, and any details on keying materials, the lifetime of timed security associations if applicable, etc.

* 网络层安全参数:这包括如何建立IPsec安全关联、应使用IPsec密钥交换机制、有关密钥材料的任何详细信息、定时安全关联的生存期(如果适用)等。

* Transport-layer security parameters: this covers how TLS connections should be established, as described in Section 5.

* 传输层安全参数:这包括如何建立TLS连接,如第5节所述。

A.2. Summary of Parameters for Consideration in Session Peering Policies

A.2. 会话对等策略中要考虑的参数摘要

The following is a summary of the parameters mentioned in the previous section. They may be part of a session peering policy and appear with a level of requirement (mandatory, recommended, supported, etc.).


o IP Network Connectivity (assumed, requirements out of scope of this document)

o IP网络连接(假设,要求不在本文件范围内)

o Media session parameters:

o 媒体会话参数:

* Codecs for audio, video, real time text, instant messaging media sessions

* 用于音频、视频、实时文本、即时消息媒体会话的编解码器

* Modes of communications for audio (voice, fax, DTMF), IM (page mode, MSRP)

* 音频(语音、传真、DTMF)、IM(页面模式、MSRP)的通信模式

* Media transport and means to establish secure media sessions

* 媒体传输和建立安全媒体会话的方法

* List of ingress and egress DBEs where applicable, including STUN Relay servers if present

* 入口和出口DBE列表(如适用),包括STUN中继服务器(如有)


o 小口喝

* SIP RFCs, methods and error responses

* SIP RFC、方法和错误响应

* headers and header values

* 标题和标题值

* possibly, list of SIP RFCs supported by groups (e.g., by call feature)

* 可能,由组支持的SIP RFC列表(例如,通过呼叫功能)

o Accounting

o 会计

o Capacity Control and Performance Management: any limits on, or, means to measure and limit the maximum number of active calls to a peer or federation, maximum number of sessions and messages per specified unit time, maximum number of active users or subscribers per specified unit time, the aggregate media bandwidth per peer or for the federation, specified SIP signaling performance metrics to measure and report; media-level VoIP metrics if applicable.

o 容量控制和性能管理:对对等方或联合体的最大活动呼叫数、每指定单位时间的最大会话数和消息数、每指定单位时间的最大活动用户数或订阅者数、每对等方或联合体的聚合媒体带宽的任何限制或手段,要测量和报告的指定SIP信令性能指标;媒体级VoIP指标(如适用)。

o Security: Call admission control, call authorization, network and transport layer security parameters, media security parameters

o 安全性:呼叫接纳控制、呼叫授权、网络和传输层安全参数、媒体安全参数

Author's Address


Jean-Francois Mule CableLabs 858 Coal Creek Circle Louisville, CO 80027 USA

Jean-Francois Mule CableLabs 858美国科罗拉多州路易斯维尔市煤溪圈80027