2024年1月19日发(作者：)
SIP协议的各字段分析和各种功能的应用
一、SIP字段说明
1．Request-Line
Request-Line = Method SP Request-URI SP SIP-Version
包含method名字、一个Request-URI和一个用空格隔开的协议版本号。Method包含SIP协议的6种命令：Register、Invite、ACK、Cancel、BYE和Options；Request-URI：标注被叫的号码或者是名字以及服务器的地址或者是URI地址；SIP-Version：在请求消息和应答消息中都包含有SIP的协议版本号，一般是“SIP/2.0”。
2．Responses
Status-Line = SIP-Version SP Status-Code SP Reason-Phrase
应答命令是用来回应请求命令的，状态行包含协议版本、状态码和原因描述（用文本方式描述状态）；状态码有1XX、2XX、3XX、4XX、5XX和6XX等六种，分别描述不同的内容。
3．Header Fields
Via：用来标识应答消息的返回路径，包含SIP版本、通过的路径、branch；在应答包中会看到此包经过别的路径后加进去的路径参数，如果是经过了NAT，那么里面会加上received=ip,rport=port这类的参数
Max-Forwards：用来表示这个包最多可以被传送多少跳，当此值为0还没有到达目的地时，系统会回483（Too Many Hops）；一般在含有request的源包里面带有这个字段，默认值正常的都是设置为70
User-Agent：终端设备自己填写的关于设备方面的信息，没有具体用处
From：源的display name和源的URI；&前面带的是设备号或者是主叫号码，&后面带的是proxy的地址
To：目的地的绝对地址，包含被叫display name和被叫URI；&前面带的是设备号或者是被叫号码，&后面带的是proxy的地址
Call-ID：是一个唯一的标识符，在一通呼叫中请求和应答消息中的Call-ID是唯一的
Contact：包含源的URI信息，用来给响应消息直接和源建立连接用
CSeq：用来标识命令和命令顺序，用32位无符号整数来表示，要小于2**31
Expires：在注册包和注册应答包中携带，用来协商URI的有效时间，当为0的时候表示注销设备，没有Expires的时候会被系统拒绝（现在做了修改，如果没有这个字段，系统会把此值当作是3600）
Allow：允许的命令
Supported：支持的操作，例如：replace等
Content-Type：指明消息体的类型
Content-Length：指明消息体的字节大小
4．Message body
SDP协议版本
会话名，例如：SIP CALL
Connection Information：Connect Network Type现在只有IN（Internet）；Connection Address
Type 现在只有IP4和IP6两种；Connection Address可以为单播地址或多播组地址，如果是多播组地址还必须有一个生存时间（TTL）值，如：c＝IN IP4 224.2.1.1/127，表示TTL＝127秒
Time Description：标明会话的开始时间和终止时间
Media Description：包含媒体类型、端口、传送层和格式列表4部分，传送层描述的是Media Protocol，例如：RTP/AVP表示IETF RTP协议，udp表示UDP协议；格式列表列举出支持的codec类型
Media Attribute：有两种描述方式：a=<属性>或a=<属性>：<值>，第一种形式称为特殊属性，无须规定数值，如：a=recvonly表示该媒体信道是“只收”信道；第二种形式称为数值属性，例如：Media Attribute Value：18 G729/8000这种形式
二、一些功能信令流程
1．Refer（Transfer业务）命令的使用
对于SIP信令的Transfer应用，主要用到的命令就是Refer，在B挂机后会向系统发出Refer命令，系统回应一个Accepted（202）命令，然后系统再会一个Notify命令，B回应200 OK后，过程结束，具体的信令流程和信令内容请查看SIP Transfer文档，下面列举如下：
Attended Call Transfer
Message 1
REFER sip:b@ SIP/2.0
Via: SIP/2.0/UDP ;branch=z9hG4bK2293940223
To:
From:
Call-ID: 898234234@
CSeq: 93809823 REFER
Max-Forwards: 70
Refer-To: (whatever URI)
Contact: sip:a@
Content-Length: 0
Message 2
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP ;branch=z9hG4bK2293940223
To:
From:
Call-ID: 898234234@
CSeq: 93809823 REFER
Contact: sip:b@
Content-Length: 0
Message 3
NOTIFY sip:a@ SIP/2.0
Via: SIP/2.0/UDP ;branch=z9hG4bK9922ef992-25
To:
From:
Call-ID: 898234234@
CSeq: 1993402 NOTIFY
Max-Forwards: 70
Event: refer
Subscription-State: active;expires=(depends on Refer-To URI)
Contact: sip:b@
Content-Type: message/sipfrag;version=2.0 普通呼叫的type为：application/sdp
Content-Length: 20
2．Replace的使用
过程如下图所示：
3．Early Media Applications
在SIP的Early Media中使用的是183信令来实现，如下图：
4． SIP自动callback的应用
5． 用Instant Message to Transfer a Call
6． SIP Message Waiting
7． SIP Call Control in Conferencing
三、 新功能
1．SigComp
the Signaling Compression (SigComp) add-on module is a solution for compressing SIP
signaling. Since SIP messages are text based, they are not optimized in terms of size. For example,
typical SIP messages range from a few hundred bytes to two thousand bytes or more (RFC 3261).
With the planned usage of these protocols in wireless and cellular handsets, and as part of the 3GPP
(Third Generation Partnership Project) requirements for IMS (IP Multimedia Subsystem), the large
message size is problematic and message compression is mandatory.
2．DLA (Dynamic Local Address)
enables the dynamic opening and closing of the local IP
addresses, which are used for request sending and reception, at any moment of the SIP Stack life
cycle. This feature is typically used with multihomed host. DLA is used for broadband (DSL, cable),
as well as wireless and cellular networks, to support handoffs and IP address changes by service
providers.
3．IP_TOS
determines the value of the Type of Service field in the IP header of all packets that are
sent over the outgoing connections. (Supported for IPv4 addresses only.)
4．Transmitter Object
the SIP Stack uses transmitter objects for message sending. Each transaction
object holds a single transmitter object and uses it to send SIP messages and message
retransmissions. The application can use this new object for sending an out-of-scope request or
response without using the Transaction layer.
5．REFER RFC 3515
REFER is a SIP method defined by RFC 3515 (Session Initiation Protocol
Refer Method). The REFER method indicates that the recipient of the REFER request should contact
a third party using the contact information provided in the REFER request. RFC 3515 provides a
mechanism allowing the party that is sending the REFER to be notified of the outcome of the
referenced request with a NOTIFY request. This implementation uses subscription objects for
REFER implementation. It replaces the previous REFER implementation and is introduced due to
standard modifications.
6．Client-side forking
in previous SIP Toolkit versions, if a request was forked, the SIP Stack on the
client side automatically connected the first established dialog, and did not allow the application to
choose a different dialog or connect multiple dialogs. In the current version, the Call-leg and
Subscription modules were enhanced to allow this flexibility.
7．Merging disabling
if a proxy forks a request and eventually the two requests are terminated at the same User Agent Server
(UAS), the UAS needs to merge them into one request. There are cases where the UAS is a gateway
and therefore will want to avoid the message merging. This flexibility is currently available.
8．Transport Layer enhancements
two functionalities were added. An application can now block incoming connections even before data
was received on them .This lets the application implement a white/black IP address list and handle the
denial of service attacks in a better way. Access to the incoming raw buffer so that the application can
dump buffer to file or discard the buffer, for example.
9．A-synchronous DNS
DNS functionality was enhanced and is now a-synchronous, improving performance especially for
multi-session applications such as gateways and servers.
10．DNS server runtime configuration
in some cases, it is required to change the DNS server that is being used at runtime. This is now possible
via the SIP Toolkit APIs.
11．Manual PRACK
the application can now control the sending of PRACK/200. This is an addition to the automatic
functionality available in previous versions.
12．Primitives compilation flag
this new compilation flag replaces the EXTRA_LEAN compilation flag. It removes the dialog layer,
allowing application to work directly above the Transaction layer. Additionally, it removes specific support
of certain headers. The application can still use these headers by adding specific support in the
application itself.
13．Middle Layer for low-level services
the RADVISION operating system abstraction layer was wrapped and some of its APIs are now exposed
so the application can use services such as timers and select.
14．Via header control
some implementations, such as when using a STUN/TURN server, require manual modification of the
Via header after DNS was completed. This control and flexibility is now possible.
15．Subscription high availability
the RADVISION SIP Stack provides a store and restore mechanism that enables the application to back
up subscriptions in the ACTIVE state. Backing up subscriptions lets application developers implement
redundancy capabilities in their systems, allowing back-up systems to “take over” when the primary
system goes down. When storing an active subscription, all of the subscription parameters are copied
into a consecutive buffer. The application can then save this buffer and use it when restoring the backup
object.
16．Authentication enhancement
the authentication mechanism enables a User Agent Client (UAC) to prove authenticity to servers or
proxies which require authentication. The SIP Stack supports SIP authentication using the HTTP Digest
Scheme as described in RFC 3261 and RFC 2617. In the current version, support of the “auth-int” quality
of protection (qop) was added. For server-side authentication, only “auth” qop is supported.
17．Requirements for End-to-Middle Security for the Session
Initiation Protocol (SIP) 【rfc4189】
A Session Initiation Protocol (SIP) User Agent (UA) does not always
trust all intermediaries in its request path to inspect its message
bodies and/or headers contained in its message. The UA might want to
protect the message bodies and/or headers from intermediaries, except
those that provide services based on its content. This situation
requires a mechanism called "end-to-middle security" to secure the
information passed between the UA and intermediaries, which does not
interfere with end-to-end security. This document defines a set of
requirements for a mechanism to achieve end-to-middle security
18．The Stream Control Transmission Protocol (SCTP) as a
Transport for the Session Initiation Protocol (SIP)
【rfc4168】
This document specifies a mechanism for usage of SCTP (the Stream
Control Transmission Protocol) as the transport mechanism between SIP
(Session Initiation Protocol) entities. SCTP is a new protocol that
provides several features that may prove beneficial for transport
between SIP entities that exchange a large amount of messages,
including gateways and proxies. As SIP is transport-independent,
support of SCTP is a relatively straightforward process, nearly
identical to support for TCP
19．Session Initiation Protocol (SIP)-H.323 Interworking
Requirements 【rfc4123】
This document describes the requirements for the logical entity known
as the Session Initiation Protocol (SIP)-H.323 Interworking Function
(SIP-H.323 IWF) that will allow the interworking between SIP and
H.323
20．Transcoding Services Invocation in the Session Initiation
Protocol (SIP)Using Third Party Call Control (3pcc)【rfc4117】
This document describes how to invoke transcoding services using
Session Initiation Protocol (SIP) and third party call control. This
way of invocation meets the requirements for SIP regarding transcoding
services invocation to support deaf, hard of hearing and speech-impaired
individuals
21．Usage of the Session Description Protocol
(SDP)Alternative Network Address Types (ANAT)
Semantics in the Session Initiation Protocol (SIP)【rfc4092】
This document describes how to use the Alternative Network Address
Types (ANAT) semantics of the Session Description Protocol (SDP)
grouping framework in SIP. In particular, we define the sdp-anat SIP
option-tag. This SIP option-tag ensures that SDP session descriptions that
use ANAT are only handled by SIP entities with ANAT support. To justify
the need for such a SIP option-tag, we describe what could possibly happen
if an ANAT-unaware SIP entity tried to handle media lines grouped with
ANAT
22．Input 3rd-Generation Partnership Project (3GPP)
Release 5 Requirements on the Session Initiation Protocol
(SIP) 【rfc4083】
The 3rd-Generation Partnership Project (3GPP) has selected Session
Initiation Protocol (SIP) as the session establishment protocol for
the 3GPP IP Multimedia Core Network Subsystem (IMS). IMS is part of
Release 5 of the 3GPP specifications. Although SIP is a protocol
that fulfills most of the requirements for establishing a session in
an IP network, SIP has never been evaluated against the specific 3GPP
requirements for operation in a cellular network. In this document,
we express the requirements identified by 3GPP to support SIP for
Release 5 of the 3GPP IMS in cellular networks
23．Update to the Session Initiation Protocol
(SIP)Preconditions Framework【rfc4032】
This document updates RFC 3312, which defines the framework for
preconditions in SIP. We provide guidelines for authors of new
precondition types and describe how to use SIP preconditions in
situations that involve session mobility
24. Interworking SIP and Intelligent Network (IN)
Applications【rfc3976】
Public Switched Telephone Network (PSTN) services such as 800-number
routing (freephone), time-and-day routing, credit-card calling, and
virtual private network (mapping a private network number into a
public number) are realized by the Intelligent Network (IN). This
document addresses means to support existing IN services from Session
Initiation Protocol (SIP) endpoints for an IP-host-to-phone call.
The call request is originated on a SIP endpoint, but the services to
the call are provided by the data and procedures resident in the
PSTN/IN. To provide IN services in a transparent manner to SIP
endpoints, this document describes the mechanism for interworking SIP
and Intelligent Network Application Part (INAP)
25．The Internet Assigned Number Authority (IANA)
Uniform Resource Identifier (URI) Parameter Registry for
the Session Initiation Protocol (SIP) 【rfc3969】
This document creates an Internet Assigned Number Authority (IANA)
registry for the Session Initiation Protocol (SIP) and SIPS Uniform
Resource Identifier (URI) parameters, and their values. It also
lists the already existing parameters to be used as initial values
for that registry
26．The Internet Assigned Number Authority (IANA)
Header Field Parameter Registry for the Session Initiation
Protocol (SIP)【rfc3968】
This document creates an Internet Assigned Number Authority (IANA)
registry for the Session Initiation Protocol (SIP) header field
parameters and parameter values. It also lists the already existing
parameters and parameter values to be used as the initial entries for
this registry
27．Early Media and Ringing Tone Generation in the Session
Initiation Protocol (SIP) 【rfc3960】
This document describes how to manage early media in the Session
Initiation Protocol (SIP) using two models: the gateway model and the
application server model. It also describes the inputs one needs to
consider in defining local policies for ringing tone generation
28．The Session Initiation Protocol (SIP) "Join" Header【rfc3911】
This document defines a new header for use with SIP multi-party
applications and call control. The Join header is used to logically
join an existing SIP dialog with a new SIP dialog. This primitive
can be used to enable a variety of features, for example: "Barge-In",
answering-machine-style "Message Screening" and "Call Center Monitoring".
Note that definition of these example features is non-normative
29．Session Initiation Protocol (SIP) Extension for Event
State Publication【rfc3903】
This document describes an extension to the Session Initiation
Protocol (SIP) for publishing event state used within the SIP Events
framework. The first application of this extension is for the
publication of presence information. The mechanism described in this
document can be extended to support publication of any event state for
which there exists an appropriate event package. It is not intended to
be a general-purpose mechanism for transport of arbitrary data, as there
are better-suited mechanisms for this purpose
30．Session Initiation Protocol (SIP) Authenticated Identity
Body (AIB) Format【rfc3893】
RFC 3261 introduces the concept of adding an S/MIME body to a Session
Initiation Protocol (SIP) request or response in order to provide
reference integrity over its headers. This document provides a more
specific mechanism to derive integrity and authentication properties
from an ’authenticated identity body’, a digitally-signed SIP
message, or message fragment. A standard format for such bodies
(known as Authenticated Identity Bodies, or AIBs) is given in this
document. Some considerations for the processing of AIBs by
recipients of SIP messages with such bodies are also given
31．The Session Initiation Protocol (SIP) Referred-By
Mechanism【rfc3892】
The Session Initiation Protocol (SIP) REFER method provides a
mechanism where one party (the referrer) gives a second party (the
referee) an arbitrary URI to reference. If that URI is a SIP URI,
the referee will send a SIP request, often an INVITE, to that URI
(the refer target). This document extends the REFER method, allowing
the referrer to provide information about the REFER request to the
refer target using the referee as an intermediary. This information
includes the identity of the referrer and the URI to which the referrer
referred. The mechanism utilizes S/MIME to help protect this information
from a malicious intermediary. This protection is optional, but a recipient
may refuse to accept a request unless it is present
32．The Session Initiation Protocol (SIP) "Replaces" Header
【rfc3891】
This document defines a new header for use with Session Initiation
Protocol (SIP) multi-party applications and call control. The
Replaces header is used to logically replace an existing SIP dialog
with a new SIP dialog. This primitive can be used to enable a
variety of features, for example: "Attended Transfer" and "Call
Pickup". Note that the definition of these example features is non-
Normative
33．S/MIME Advanced Encryption Standard (AES)
Requirement for the Session Initiation Protocol (SIP)【rfc3853】
RFC 3261 currently specifies 3DES as the mandatory-to-implement
ciphersuite for implementations of S/MIME in the Session Initiation
Protocol (SIP). This document updates the normative guidance of RFC
3261 to require the Advanced Encryption Standard (AES) for S/MIME
Core SIP Documents
RFC 2543
RFC 3261
SIP: Session Initiation Protocol (obsolete)
SIP: Session Initiation Protocol
SDP-Related Documents
RFC
2327
RFC
3264
RFC
3266
RFC
3388
Session Description Protocol (SDP)
An Offer/Answer Model with the Session Description Protocol (SDP)
Support of IPv6 in SDP
Grouping Media Lines in SDP
RFC
3407
RFC
3556
RFC
3605
RFC
3890
RFC
4091
RFC
4145
Session Description Protocol (SDP) Simple Capability Declaration
SDP Bandwidth Modifiers for RTCP Bandwidth
Real Time Control Protocol (RTCP) attribute in Session Description Protocol
(SDP)
A Transport Independent Bandwidth Modifier
An Alternative NAT Semantics for SDP
TCP-Based Media Transport in the SDP
RTP-Related Documents
RFC 3550
RTP: Transport Protocol for Real-Time Applications
RFC 3551
RTP Profile for A/V Conferences with Minimal Control
RFC 2198
RTP Payload for Redundant Audio Data
RFC 2733
An RTP Payload Format for Generic Forward Error Correction
RFC 2793
RTP Payload for Text Conversation
RFC 2833
RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2959
Real-Time Transport Protocol Management Information Base
RFC 3389
RTP Payload for Comfort Noise
RFC 3611
RTP Control Protocol Extended Reports (RTCP XR)
RFC 3711
The Secure Real-time Transport Protocol (SRTP)
HTTP-Related Documents
RFC
2616
RFC
2617
RFC
3310
RFC
3310
Hypertext Transfer Protocol -- HTTP/1.1
HTTP Authentication: Basic and Digest Access Authentication
HTTP Digest Authentication Using Authentication and Key Agreement
(AKA)
HTTP Digest Authentication Using Authentication and Key Agreement
(AKA)
MIME-Related Documents
RFC 1847
Security Multiparts for MIME: Multipart/Signed and Multipart/Encrypted
RFC 2045
MIME Part One: Format of Internet Message Bodies
RFC 2046
MIME Part Two: Media Types
RFC 2047
MIME Part Three: Message Header Extensions for Non-ASCII Text
RFC 2048
MIME Part Four: Registration Procedures
RFC 2633
S/MIME Version 3 Message Specification
RFC 3204
MIME media types for ISUP and QSIG Objects
RFC 3420
Internet Media Type message/sipfrag
RFC 3555
MIME Type Registration of RTP Payload Formats
SIP Extension and Options
RFC 2976
The SIP INFO Method
RFC 2848
Extensions for IP Access to Telephone Call Services
RFC 3050
CGI for SIP
RFC 3262
Reliability of Provisional Responses
RFC 3263
Locating SIP Servers
RFC 3265
SIP-Specific Event Notification
RFC 3311
UPDATE Method
RFC 3312
Integration of Resource Management and SIP
RFC 3313
Private SIP Extensions for Media Authorization
RFC 3319
DHCPv6 Options for SIP Servers
RFC 3323
A Privacy Mechanism for SIP
RFC 3324
Short Term Requirements for Network Asserted Identity
RFC 3325
Private Extensions to SIP for Asserted Identity within Trusted Networks
RFC 3326
The Reason Header Field
RFC 3327
Extension Header Field for Registering Non-Adjacent Contacts
RFC 3329
Security Mechanism Agreement
RFC 3361
DHCP-for-IPv4 Option for SIP Servers
RFC 3372
SIP for Telephones (SIP-T): Context and Architectures
RFC 3398
ISUP to SIP Mapping
RFC 3428
SIP Extension for Instant Messaging
RFC 3455
Private Header Extensions for 3GPP
RFC 3515
The Session Initiation Protocol (SIP) Refer Method
RFC 3608
Extension Header Field for Service Route Discovery During Registration
RFC 3680
SIP Event Package for Registrations
RFC 3840
Indicating User Agent Capabilities in SIP
RFC 3841
Caller Preferences for SIP
RFC 3842
Message Summary and Message Waiting Indication Event Package
RFC 3856
Presence Event Package
RFC 3857
A Watcher Information Event Template-Package
RFC 3891
"Replaces" Header
RFC 3903
Event State Publication
RFC 3959
Early Session Disposition Type
RFC 3960
Early Media and Ringing Tone Generation
RFC 4028
Session Timers in the Session Initiation Protocol (SIP)
RFC 4235
An INVITE-Initiated Dialog Event Package for SIP
RFC 4244
Extension for Request History Information
SIP Informational RFCs and BCP Documents
RFC
3087
RFC
3351
RFC
3603
RFC
3702
RFC
3824
RFC
3911
RFC
3968
RFC
3969
RFC
3976
RFC
4117
RFC
4123
RFC
Control of Service Context using SIP Request-URI
User Requirements for SIP in Support of Speech/Hearing Impaired
Private SIP Proxy-to-Proxy Extensions for PacketCable Distributed Call
Signaling
Authentication, Authorization, and Accounting Requirements for SIP
Using E.164 numbers with SIP
The SIP "Join" Header
IANA Registry for SIP Header Field
IANA Registry for SIP URI
Interworking SIP and IN Applications
Transcoding Services Invocation using 3PCC
SIP-H.323 Interworking Requirements
SCTP as a transport for SIP
4168
RFC
4189
RFC
4245
Requirements for End-to-Middle Security for SIP
High-Level Requirements for Tightly Coupled SIP Conferencing
SIP-Related Documents
RFC 3219
Telephony Routing over IP (TRIP) (tutorial)
RFC 3320
Signalling Compression
RFC 3321
Signalling Compression - Extended Operations (informational)
RFC 3322
Signalling Compression - Requirements and Assumptions (informational)
RFC 3486
Compressing the Session Initiation Protocol (SIP)
RFC 3485
SIP and SDP Static Dictionary for Signaling Compression
RFC 3725
Best Current Practices for 3PCC in SIP
RFC 3764
enumservice registration for SIP Addresses-of-Record
RFC 4077
A Negative Acknowledgement Mechanism for Signaling Compression
RFC 4083
3GPP Release 5 Requirements on SIP
RFC 4092
Using SDP Alternative NAT Semantics in SIP
Directory Services Documents
H.350
H.350.4
Directory Services Architecture for Multimedia Conferencing
Directory Services Architecture for SIP
Online Resources for SIP
Henning Schulzrinne’s SIP Page
/sip
and supplementary SIP and SIPPING archives:
/sipwg/drafts/
/sipping/drafts/
IP Telephony Web Page
SIP Forum
SIP Center
SIP Products at
/sip/
Finding a draft
Finding an RFC
SIP WG
SIPPING WG
SIMPLE WG
XCON WG
MMUSIC WG
/
/
/rs/
/rs/
/rs/
/rs/
/rs/
发布评论