- •Contents
- •Foreword to the First Edition
- •Preface to the Third Edition
- •Preface to the Second Edition
- •Preface to the First Edition
- •1 SIP and the Internet
- •1.1 Signaling Protocols
- •1.2.2 Data/Link Layer
- •1.2.3 Network Layer
- •1.2.4 Transport Layer
- •1.2 Internet Multimedia Protocol Stack
- •1.2.1 Physical Layer
- •1.2.5 Application Layer
- •1.2.6 Utility Applications
- •1.2.7 Multicast
- •1.3 Internet Names
- •1.4 URLs, URIs, and URNs
- •1.5 Domain Name Service
- •1.5.1 DNS Resource Records
- •1.5.2 Address Resource Records (A or AAAA)
- •1.5.3 Service Resource Records (SRV)
- •1.5.4 Naming Authority Pointer Resource Records (NAPTR)
- •1.5.5 DNS Resolvers
- •1.6 Global Open Standards
- •1.7 Internet Standards Process
- •1.8 A Brief History of SIP
- •1.9 Conclusion
- •References
- •2 Introduction to SIP
- •2.1 A Simple Session Establishment Example
- •2.2 SIP Call with a Proxy Server
- •2.3 SIP Registration Example
- •2.4 SIP Presence and Instant Message Example
- •2.5 Message Transport
- •2.5.1 UDP Transport
- •2.5.2 TCP Transport
- •2.5.3 TLS Transport
- •2.5.4 SCTP Transport
- •2.6 Transport Protocol Selection
- •2.7 Conclusion
- •2.8 Questions
- •References
- •3 SIP Clients and Servers
- •3.1 SIP User Agents
- •3.2 Presence Agents
- •3.3 Back-to-Back User Agents
- •3.4 SIP Gateways
- •3.5 SIP Servers
- •3.5.1 Proxy Servers
- •3.5.2 Redirect Servers
- •3.5.3 Registrar Servers
- •3.6 Uniform Resource Indicators
- •3.7 Acknowledgment of Messages
- •3.8 Reliability
- •3.9 Multicast Support
- •3.10 Conclusion
- •3.11 Questions
- •References
- •4 SIP Request Messages
- •4.1 Methods
- •4.1.1 INVITE
- •4.1.2 REGISTER
- •4.1.5 CANCEL
- •4.1.6 OPTIONS
- •4.1.7 SUBSCRIBE
- •4.1.8 NOTIFY
- •4.1.9 PUBLISH
- •4.1.10 REFER
- •4.1.11 MESSAGE
- •4.1.12 INFO
- •4.1.13 PRACK
- •4.1.14 UPDATE
- •4.2 URI and URL Schemes Used by SIP
- •4.2.1 SIP and SIPS URIs
- •4.2.2 Telephone URLs
- •4.2.3 Presence and Instant Messaging URLs
- •4.3 Tags
- •4.4 Message Bodies
- •4.5 Conclusion
- •4.6 Questions
- •References
- •5 SIP Response Messages
- •5.1 Informational
- •5.1.1 100 Trying
- •5.1.2 180 Ringing
- •5.1.3 181 Call is Being Forwarded
- •5.1.4 182 Call Queued
- •5.1.5 183 Session Progress
- •5.2 Success
- •5.2.2 202 Accepted
- •5.3 Redirection
- •5.3.1 300 Multiple Choices
- •5.3.2 301 Moved Permanently
- •5.3.3 302 Moved Temporarily
- •5.3.4 305 Use Proxy
- •5.3.5 380 Alternative Service
- •5.4.3 402 Payment Required
- •5.4.4 403 Forbidden
- •5.4.5 404 Not Found
- •5.4.6 405 Method Not Allowed
- •5.4.7 406 Not Acceptable
- •5.4.8 407 Proxy Authentication Required
- •5.4.9 408 Request Timeout
- •5.4.11 410 Gone
- •5.4.12 411 Length Required
- •5.4.13 412 Conditional Request Failed
- •5.4 Client Error
- •5.4.1 400 Bad Request
- •5.4.2 401 Unauthorized
- •5.4.14 413 Request Entity Too Large
- •5.4.15 414 Request-URI Too Long
- •5.4.16 415 Unsupported Media Type
- •5.4.17 416 Unsupported URI Scheme
- •5.4.18 417 Unknown Resource Priority
- •5.4.19 420 Bad Extension
- •5.4.20 421 Extension Required
- •5.4.21 422 Session Timer Interval Too Small
- •5.4.22 423 Interval Too Brief
- •5.4.23 428 Use Identity Header
- •5.4.24 429 Provide Referror Identity
- •5.4.25 430 Flow Failed
- •5.4.26 433 Anonymity Disallowed
- •5.4.27 436 Bad Identity-Info Header
- •5.4.29 438 Invalid Identity Header
- •5.4.30 439 First Hop Lacks Outbound Support
- •5.4.31 440 Max-Breadth Exceeded
- •5.4.32 470 Consent Needed
- •5.4.33 480 Temporarily Unavailable
- •5.4.34 481 Dialog/Transaction Does Not Exist
- •5.4.35 482 Loop Detected
- •5.4.36 483 Too Many Hops
- •5.4.37 484 Address Incomplete
- •5.4.38 485 Ambiguous
- •5.4.39 486 Busy Here
- •5.4.40 487 Request Terminated
- •5.4.41 488 Not Acceptable Here
- •5.4.42 489 Bad Event
- •5.4.43 491 Request Pending
- •5.4.44 493 Request Undecipherable
- •5.4.45 494 Security Agreement Required
- •5.5 Server Error
- •5.5.1 500 Server Internal Error
- •5.5.2 501 Not Implemented
- •5.5.3 502 Bad Gateway
- •5.5.4 503 Service Unavailable
- •5.5.5 504 Gateway Timeout
- •5.5.6 505 Version Not Supported
- •5.5.7 513 Message Too Large
- •5.5.8 580 Preconditions Failure
- •5.6 Global Error
- •5.6.1 600 Busy Everywhere
- •5.6.2 603 Decline
- •5.6.3 604 Does Not Exist Anywhere
- •5.6.4 606 Not Acceptable
- •5.7 Questions
- •References
- •6 SIP Header Fields
- •6.1 Request and Response Header Fields
- •6.1.1 Accept
- •6.1.2 Accept-Encoding
- •6.1.3 Accept-Language
- •6.1.4 Alert-Info
- •6.1.5 Allow
- •6.1.6 Allow-Events
- •6.1.7 Answer-Mode
- •6.1.8 Call-ID
- •6.1.9 Contact
- •6.1.10 CSeq
- •6.1.11 Date
- •6.1.12 Encryption
- •6.1.13 Expires
- •6.1.14 From
- •6.1.15 History Info
- •6.1.16 Organization
- •6.1.17 Path
- •6.1.19 Record-Route
- •6.1.20 Recv-Info
- •6.1.21 Refer-Sub
- •6.1.22 Retry-After
- •6.1.23 Subject
- •6.1.24 Supported
- •6.1.25 Timestamp
- •6.1.27 User-Agent
- •6.2 Request Header Fields
- •6.2.1 Accept-Contact
- •6.2.2 Authorization
- •6.2.3 Call-Info
- •6.2.4 Event
- •6.2.5 Hide
- •6.2.6 Identity
- •6.2.7 Identity-Info
- •6.2.8 In-Reply-To
- •6.2.9 Info-Package
- •6.2.10 Join
- •6.2.11 Priority
- •6.2.12 Privacy
- •6.2.13 Proxy-Authorization
- •6.2.14 Proxy-Require
- •6.2.15 P-OSP-Auth-Token
- •6.2.16 P-Asserted-Identity
- •6.2.17 P-Preferred-Identity
- •6.2.18 Max-Breadth
- •6.2.19 Max-Forwards
- •6.2.20 Reason
- •6.2.21 Refer-To
- •6.2.22 Referred-By
- •6.2.23 Reply-To
- •6.2.24 Replaces
- •6.2.25 Reject-Contact
- •6.2.26 Request-Disposition
- •6.2.27 Require
- •6.2.28 Resource-Priority
- •6.2.29 Response-Key
- •6.2.30 Route
- •6.2.31 RAck
- •6.2.32 Security-Client
- •6.2.33 Security-Verify
- •6.2.34 Session-Expires
- •6.2.35 SIP-If-Match
- •6.2.36 Subscription-State
- •6.2.37 Suppress-If-Match
- •6.2.38 Target-Dialog
- •6.2.39 Trigger-Consent
- •6.3 Response Header Fields
- •6.3.1 Accept-Resource-Priority
- •6.3.2 Authentication-Info
- •6.3.3 Error-Info
- •6.3.4 Flow-Timer
- •6.3.5 Min-Expires
- •6.3.7 Permission-Missing
- •6.3.8 Proxy-Authenticate
- •6.3.9 Security-Server
- •6.3.10 Server
- •6.3.11 Service-Route
- •6.3.12 SIP-ETag
- •6.3.13 Unsupported
- •6.3.14 Warning
- •6.3.15 WWW-Authenticate
- •6.3.16 RSeq
- •6.4 Message Body Header Fields
- •6.4.1 Content-Encoding
- •6.4.2 Content-Disposition
- •6.4.3 Content-Language
- •6.4.4 Content-Length
- •6.4.5 Content-Type
- •6.4.6 MIME-Version
- •6.5 Questions
- •References
- •7 Wireless, Mobility, and IMS
- •7.1 IP Mobility
- •7.2 SIP Mobility
- •7.4 IMS Header Fields
- •7.5 Conclusion
- •7.6 Questions
- •References
- •8 Presence and Instant Messaging
- •8.1 Introduction
- •8.2 History of IM and Presence
- •8.3 SIMPLE
- •8.4 Presence with SIMPLE
- •8.4.1 SIP Events Framework
- •8.4.2 Presence Bodies
- •8.4.3 Resource Lists
- •8.4.4 Filtering
- •8.4.6 Partial Publication
- •8.4.7 Presence Documents Summary
- •8.5 Instant Messaging with SIMPLE
- •8.5.1 Page Mode Instant Messaging
- •8.5.4 Message Composition Indication
- •8.5.5 Multiple Recipient Messages
- •8.5.6 Session Mode Instant Messaging
- •8.6 Jabber
- •8.6.1 Standardization as Extensible Messaging and Presence Protocol
- •8.6.2 Interworking with SIMPLE
- •8.6.3 Jingle
- •8.6.4 Future Standardization of XMPP
- •8.7 Conclusion
- •8.8 Questions
- •References
- •9 Services in SIP
- •9.1 Gateway Services
- •9.2 SIP Trunking
- •9.3 SIP Service Examples
- •9.4 Voicemail
- •9.5 SIP Video
- •9.6 Facsimile
- •9.7 Conferencing
- •9.7.1 Focus
- •9.7.2 Mixer
- •9.8 Application Sequencing
- •9.9 Other SIP Service Architectures
- •9.9.1 Service Oriented Architecture
- •9.9.2 Servlets
- •9.9.3 Service Delivery Platform
- •9.10 Conclusion
- •9.11 Questions
- •References
- •10 Network Address Translation
- •10.1 Introduction to NAT
- •10.2 Advantages of NAT
- •10.3 Disadvantages of NAT
- •10.4 How NAT Works
- •10.5 Types of NAT
- •10.5.1 Endpoint Independent Mapping NAT
- •10.5.2 Address Dependent Mapping NAT
- •10.5.3 Address and Port Dependent Mapping NAT
- •10.5.4 Hairpinning Support
- •10.5.5 IP Address Pooling Options
- •10.5.6 Port Assignment Options
- •10.5.7 Mapping Refresh
- •10.5.8 Filtering Modes
- •10.6 NAT Mapping Examples
- •10.7 NATs and SIP
- •10.8 Properties of a Friendly NAT or How a NAT Should BEHAVE
- •10.9 STUN Protocol
- •10.10 UNSAF Requirements
- •10.11 SIP Problems with NAT
- •10.11.1 Symmetric SIP
- •10.11.2 Connection Reuse
- •10.11.3 SIP Outbound
- •10.12 Media NAT Traversal Solutions
- •10.12.1 Symmetric RTP
- •10.12.2 RTCP Attribute
- •10.12.3 Self-Fixing Approach
- •10.13 Hole Punching
- •10.14 TURN: Traversal Using Relays Around NAT
- •10.15 ICE: Interactive Connectivity Establishment
- •10.16 Conclusion
- •10.17 Questions
- •References
- •11 Related Protocols
- •11.1 PSTN Protocols
- •11.1.1 Circuit Associated Signaling
- •11.1.2 ISDN Signaling
- •11.1.3 ISUP Signaling
- •11.2 SIP for Telephones
- •11.3 Media Gateway Control Protocols
- •11.4.1 Introduction to H.323
- •11.4.2 Example of H.323
- •11.4.3 Versions
- •References
- •12 Media Transport
- •12.1 Real-Time Transport Protocol (RTP)
- •12.2 RTP Control Protocol (RTCP)
- •12.2.1 RTCP Reports
- •12.2.2 RTCP Extended Reports
- •12.3 Compression
- •12.4.1 Audio Codecs
- •12.4.2 Video Codecs
- •12.5 Conferencing
- •12.6 ToIP—Conversational Text
- •12.7 DTMF Transport
- •12.8 Questions
- •References
- •13 Negotiating Media Sessions
- •13.1 Session Description Protocol (SDP)
- •13.1.1 Protocol Version
- •13.1.2 Origin
- •13.1.3 Session Name and Information
- •13.1.5 E-Mail Address and Phone Number
- •13.1.6 Connection Data
- •13.1.7 Bandwidth
- •13.1.8 Time, Repeat Times, and Time Zones
- •13.1.9 Encryption Keys
- •13.1.10 Media Announcements
- •13.1.11 Attributes
- •13.2 SDP Extensions
- •13.3 The Offer Answer Model
- •13.3.1 Rules for Generating an Offer
- •13.3.2 Rules for Generating an Answer
- •13.3.3 Rules for Modifying a Session
- •13.3.4 Special Case—Call Hold
- •13.4 Static and Dynamic Payloads
- •13.5 SIP Offer Answer Exchanges
- •13.6 Conclusion
- •13.7 Questions
- •References
- •14 SIP Security
- •14.1 Basic Security Concepts
- •14.1.1 Encryption
- •14.1.2 Public Key Cryptography
- •14.1.4 Message Authentication
- •14.2 Threats
- •14.3 Security Protocols
- •14.3.1 IPSec
- •14.3.3 DNSSec
- •14.3.4 Secure MIME
- •14.4 SIP Security Model
- •14.4.1 SIP Digest Authentication
- •14.4.2 SIP Authentication Using TLS
- •14.4.3 Secure SIP
- •14.4.4 Identity
- •14.4.5 Enhanced SIP Identity
- •14.6 Media Security
- •14.6.1 Non-RTP Media
- •14.6.2 Secure RTP
- •14.6.3 Keying SRTP
- •14.6.4 Best Effort Encryption
- •14.6.5 ZRTP
- •14.7 Questions
- •References
- •15 Peer-to-Peer SIP
- •15.1 P2P Properties
- •15.2 P2P Properties of SIP
- •15.3 P2P Overlays
- •15.4 RELOAD
- •15.5 Host Identity Protocol
- •15.6 Conclusion
- •15.7 Questions
- •References
- •16 Call Flow Examples
- •16.1 SIP Call with Authentication, Proxies, and Record-Route
- •16.2 SIP Call with Stateless and Stateful Proxies with Called Party Busy
- •16.3 SIP to PSTN Call Through Gateways
- •16.4 PSTN to SIP Call Through a Gateway
- •16.5 Parallel Search
- •16.6 Call Setup with Two Proxies
- •16.7 SIP Presence and Instant Message Example
- •References
- •17 Future Directions
- •17.2 More Extensions
- •17.3 Better Identity
- •17.4 Interdomain SIP
- •17.5 Making Features Work Better
- •17.6 Emergency Calling
- •17.7 More SIP Trunking
- •17.9 Improved NAT Traversal
- •17.10 Security Deployment
- •17.11 Better Interoperability
- •References
- •A.1 ABNF Rules
- •A.2 Introduction to XML
- •References
- •About the Author
Introduction to SIP |
31 |
CSeq: 1392 BYE
Content-Length: 0
The response echoes the CSeq of the original request: 1392 BYE. No ACK is sent since ACK is only sent in response to INVITE requests.
2.2 SIP Call with a Proxy Server
In the SIP message exchange of Figure 2.1, Tesla knew the IP address of Marconi and was able to send the INVITE directly to that address. This will not be the case in general—an IP address cannot be used like a telephone number. One reason is that IP addresses are often dynamically assigned due to the shortage of IPv4 addresses. For example, when a PC dials in to an Internet service provider (ISP) modem bank, an IP address is assigned using DHCP to the PC from a pool of available addresses allocated to the ISP. For the duration of the session, the IP address does not change, but it is different for each dial-in session. Even for an “always on” Internet connection such as a DSL line, a different IP address can be assigned after each reboot of the PC. Also, an IP address does not uniquely identify a user, but identifies a node on a particular physical IP network. You have one IP address at your office, another at home, and still another when you log on remotely while traveling. Ideally, there would be one address that would identify you wherever you are. In fact, there is an Internet protocol that does exactly that with e-mail. SMTP uses a host or system independent name (an e-mail address) that does not correspond to a particular IP address. It allows e-mail messages to reach you regardless of what your IP address is and where you are logged onto the Internet.
In addition, a request routed using only IP addresses will reach only one end point—only one device. Since communication is typically user-to-user instead of device-to-device, a more useful addressing scheme would allow a particular user to call another particular user, which would result in the request reaching the target user regardless of which device they are currently using, or if they have multiple devices.
SIP uses e-mail-like names for addresses. The addressing scheme is part of a family of Internet addresses known as URIs, as described in Section 1.4. SIP URIs can also handle telephone numbers, transport parameters, and a number of other items. A full description, including examples, can be found in Section 4.2. For now, the key point is that a SIP URI is a name that is resolved to an IP address by using a SIP proxy server and DNS lookups at the time of the call, as will be seen in the next example. Figure 2.2 shows an example of a more typical SIP call with a type of SIP server called a proxy server. In this example, the caller Schroedinger calls Heisenberg through a SIP proxy server. A SIP proxy operates in a similar way to a proxy in HTTP and other Internet protocols. A SIP proxy
32 |
SIP: Understanding the Session Initiation Protocol |
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Figure 2.2 SIP call example with a proxy server.
does not set up or terminate sessions, but sits in the middle of a SIP message exchange, receiving messages and forwarding them. This example shows one proxy, but there can be multiple proxies in a signaling path.
SIP has two broad categories of URIs: ones that correspond to a user, and ones that correspond to a single device or end point. The user URI is known as an address of record (AOR), and a request sent to an address of record will require database lookups and service and feature operations, which can result in the request being sent to one or more end devices. A device URI is known as a contact, and typically does not require database lookups. An address of record URI is usually used in To and From header fields, as this is the general way to reach a person and is suitable for storing in address books and in returning missed calls. A device URI is usually used in a Contact header field and is associated with a particular user for a shorter period of time. The method of relating (or binding) a contact URI with an address of record URI will be discussed in Section 2.3.
Because Schroedinger does not know exactly where Heisenberg is currently logged on and what device he is currently using, a SIP proxy server is used to route the INVITE. First, a DNS lookup of Heisenberg’s SIP URI domain name (munich.de) is performed (see Section 2.6 for the details), which returns the IP address of the proxy server proxy.munich.de, which handles that domain. The INVITE is then sent to that IP address:
INVITE sip:werner.heisenberg@munich.de SIP/2.0
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a Max-Forwards: 70
To: Heisenberg <sip:werner.heisenberg@munich.de>
From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42
Introduction to SIP |
33 |
Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Subject: Where are you exactly?
Contact: <sip:schroed5244@pc33.wave.org>
Content-Type: application/sdp
Content-Length: 159
v=0
o=schroed5244 2890844526 2890844526 IN IP4 100.101.102.103 s=Phone Call
t=0 0
c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000
The proxy looks up the SIP URI in the Request-URI (sip:werner. heisenberg@munich.de) in its database and locates Heisenberg. This completes the two-step process of:
1.DNS lookup by user agent to locate the IP address of the proxy. Database lookup is performed by the proxy to locate the IP address.
2.The INVITE is then forwarded to Heisenberg’s IP address with the addition of a second Via header field stamped with the address of the proxy.
INVITE sip:werner.heisenberg@200.201.202.203
Via: SIP/2.0/UDP proxy.munich.de:5060;branch=z9hG4bK83842.1 Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a Max-Forwards: 69
To: Heisenberg <sip:werner.heisenberg@munich.de>
From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Contact: <sip:schroed5244@pc33.wave.org> Content-Type: application/sdp Content-Length: 159
v=0
o=schroed5244 2890844526 2890844526 IN IP4 100.101.102.103 s=Phone Call
c=IN IP4 100.101.102.103 t=0 0
m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
From the presence of two Via header fields, Heisenberg knows that the INVITE has been routed through a proxy server. Having received the INVITE, a 180 Ringing response is sent by Heisenberg to the proxy:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP proxy.munich.de:5060;branch=z9hG4bK83842.1 ;received=100.101.102.105
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a
To: Heisenberg <sip:werner.heisenberg@munich.de>;tag=314159
34 |
SIP: Understanding the Session Initiation Protocol |
From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Contact: <sip:werner.heisenberg@200.201.202.203> Content-Length: 0
Again, this response contains the Via header fields, and the To, From, Call-ID, and CSeq header fields from the INVITE request. The response is then sent to the address in the first Via header field, proxy.munich.de to the port number listed in the Via header field: 5060, in this case. Notice that the To header field now has a tag added to it to identify this particular dialog. Only the first Via header field contains a received parameter, since the second Via header already contains the literal IP address in the URI. The Contact header field contains the device URI of Heisenberg.
The proxy receives the response, checks that the first Via header field has its own address (proxy.munich.de), uses the transaction identifier in the Via header to locate the transaction, removes that Via header field, then forwards the response to the address in the next Via header field: IP address 100.101.102.103, port 5060. The resulting response sent by the proxy to Schroedinger is:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a
To: Heisenberg <sip:werner.heisenberg@munich.de>;tag=314159 From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Contact: <sip:werner.heisenberg@200.201.202.203> Content-Length: 0
The use of Via header fields in routing and forwarding SIP messages reduces complexity in message forwarding. The request required a database lookup by the proxy to be routed. The response requires no lookup because the routing is imbedded in the message in the Via header fields. This ensures that responses route back through the same set of proxies as the request. The call is accepted by Heisenberg, who sends a 200 OK response:
SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.munich.de:5060;branch=z9hG4bK83842.1 ;received=100.101.102.105
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a
To: Heisenberg <sip:werner.heisenberg@munich.de>;tag=314159 From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Contact: <sip:werner.heisenberg@200.201.202.203> Content-Type: application/sdp
Content-Length: 159
v=0
o=heisenberg 2890844526 2890844526 IN IP4 200.201.202.203 s=Phone Call
Introduction to SIP |
35 |
c=IN IP4 200.201.202.203 t=0 0
m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
The proxy forwards the 200 OK message to Schroedinger after removing the first Via header field:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKmp17a
To: Heisenberg <sip:werner.heisenberg@munich.de>;tag=314159 From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 INVITE
Contact: <sip:werner.heisenberg@200.201.202.203> Content-Type: application/sdp
Content-Length: 159
v=0
o=heisenberg 2890844526 2890844526 IN IP4 200.201.202.203 s=phone call
c=IN IP4 200.201.202.203 t=0 0
m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000
The presence of the Contact header field with the SIP URI address of Heisenberg in the 200 OK allows Schroedinger to send the ACK directly to Heisenberg, bypassing the proxy. (Note that the Request-URI is set to Heisenberg’s Contact URI and not the URI in the To header field.) This request, and all future requests, continue to use the tag in the To header field:
ACK sip:werner.heisenberg@200.201.202.203
Via: SIP/2.0/UDP 100.101.102.103:5060;branch=z9hG4bKka42 Max-Forwards: 70
To: Heisenberg <sip:werner.heisenberg@munich.de>;tag=314159 From: E. Schroedinger <sip:schroed5244@wave.org>;tag=42 Call-ID: 4827311-391-32934
CSeq: 1 ACK Content-Length: 0
This shows that the proxy server is not really “in the call.” It facilitates the two end points locating and contacting each other, but it can drop out of the signaling path as soon as it no longer adds any value to the exchange. This role of helping the two user agents locate each other is sometimes called rendezvous and is a key function of the SIP protocol. A proxy server can force further messaging to route through it by inserting a Record-Route header field, which is described in Section 6.119. In addition, it is possible to have a proxy server that does not retain any knowledge of the fact that there is a session established between Schroedinger and Heisenberg (referred to as call state information). This
