- •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 |
47 |
agent to proxy TCP connection, this is usually not a problem unless the two have many simultaneous dialogs established.
SCTP also supports multihoming, so if one of a pair of load balancing SIP proxies fails, the other can immediately begin receiving the messages without even requiring a DNS or other database lookup. The SIP usage of SCTP is described in [9].
2.6 Transport Protocol Selection
Since SIP supports multiple transport protocols, it must have a way of managing them. The full SIP usage of DNS is defined in [10], but the basic steps for a client are listed here. There are two ways that this is acheived. The first is through explicit indications in a SIP URI. The presence of a transport=tcp or transport=sctp [9] indicates that the particular transport protocol should be used for this URI. Note that for TLS transport, the SIPS URI scheme should be used, although some implementations use the nonstandard transport=tls parameter. UDP is used if transport=udp is included. When no transport parameter is included, the following rules are followed:
1.If the URI has a numeric IP address, then UDP should be used for a SIP URI and TCP for a SIPS URI.
2.If the URI does not have a numeric address but has a numeric port number, then UDP should be used for a SIP URI and TCP for a SIPS URI.
3.If the URI does not have a numeric IP address or port, and NAPTR DNS queries are supported, then a DNS NAPTR query should be performed on the host part of the URI. The NAPTR service fields are “SIP+D2U” for UDP, “SIP+D2T” for TCP, and “SIP+D2S” for SCTP transport. The result of the NAPTR regex replacement will be a URI, which is used for an SRV lookup described in the next step. The preference field indicates the relative preference if multiple transports are supported. If no NAPTR records are returned, an SRV query should be performed.
4.The SIP usage of DNS SRV records uses “_sip” or “_sips” for the protocol and “_udp”, “_tcp”, and “_sctp” for UDP, TCP, and SCTP transports. The results of the SRV query will be a target hostname and port number. The request should be sent to that address and port. Full details on SRV record usage are in [11].
48 |
SIP: Understanding the Session Initiation Protocol |
5.If no SRV records are found, then an address query for A or an AAAA DNS query should be performed, and UDP should be used for a SIP URI and TCP for a SIPS URI.
A slightly different set of rules are followed by proxy servers, as described in [10].
For example, consider the DNS lookup performed by Schroedinger in Figure 2.2. The URI is sip:werner.heisenberg@munich.de, which does not contain a numeric IP address or port, so steps 1 and 2 are not followed. Schroedinger then follows step 3 and performs a NAPTR query on munich.de which returns the following record:
munich.de. 360 IN NAPTR 100 50 “s” “SIPS+DTU” “” _sip._udp.munich.de
Since only UDP transport is supported, step 4 results in an SRV query on _sip._udp.munich.de, which returns the following record:
_sip._udp.munich.de. 300 IN SRV 0 100 5060 proxy.munich.de
Finally, an A lookup is performed on proxy.munich.de which returns:
proxy.munich.de. 3600 IN A 100.101.102.105
As a result, Schroedinger sends the INVITE to 100.101.102.105:5060 as shown in Figure 2.2.
2.7 Conclusion
This chapter introduced the Session Initiation Protocol using some common call flow examples including a basic call, call through a proxy server, registration, and presence and instant messaging. The next chapter will explore further the details of SIP and the behavior or standard elements such as user agents, proxies, redirect servers, and registrars.
2.8Questions
Q2.1 Define a SIP dialog.
Q2.2 What SIP parameter carries the SIP transaction identifier? Q2.3 Compare proxy, registrar, and redirect servers.
Introduction to SIP |
49 |
Q2.4 Which SIP methods create dialogs? Which SIP methods end dialogs?
Q2.5 Explain the purpose of the Contact header field in an INVITE.
Q2.6 Is the Content-Length header field mandatory for TCP transport? Why or why not?
Q2.7 What is the purpose of Via header fields?
Q2.8 The DNS application Dig has returned the following values. What are the three types of DNS Resource returned? Explain the meaning of each field of the record for _sip._tcp.iptel.org. What IP address and port would a SIP request (Service = sip) be sent to at the iptel.org domain, assuming TCP transport (Proto=tcp)?
$ dig _sip._tcp.iptel.org in srv
; <<>> DiG 9.3.4 <<>> _sip._tcp.iptel.org in srv
;;global options: printcmd
;;Got answer:
;;->>HEADER<<- opcode: QUERY, status: NOERROR, id: 15807
;;flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 5, ADDITIONAL: 5
;;QUESTION SECTION:
;_sip._tcp.iptel.org. IN SRV
;; ANSWER SECTION: |
|
|
|
|
_sip._tcp.iptel.org. 86400 IN |
SRV 0 0 15060 sip. |
|||
iptel.org. |
|
|
|
|
;; AUTHORITY SECTION: |
|
|
|
|
iptel.org. |
51797 |
IN |
NS |
ns4.mydyndns.org. |
iptel.org. |
51797 |
IN |
NS |
ns5.mydyndns.org. |
iptel.org. |
51797 |
IN |
NS |
ns.iptel.org. |
iptel.org. |
51797 |
IN |
NS |
ns2.mydyndns.org. |
iptel.org. |
51797 |
IN |
NS |
ns3.mydyndns.org. |
;; ADDITIONAL SECTION: |
|
|
|
|
sip.iptel.org. |
86400 IN |
|
A |
213.192.59.75 |
ns2.mydyndns.org. |
41406 |
IN |
A |
204.13.249.76 |
ns3.mydyndns.org. |
41406 |
IN |
A |
208.78.69.76 |
ns4.mydyndns.org. |
41406 |
IN |
A |
91.198.22.76 |
ns5.mydyndns.org. |
43021 |
IN |
A |
203.62.195.76 |
;;Query time: 178 msec
;;SERVER: 192.168.0.1#53(192.168.0.1)
;;WHEN: Wed Feb 6 09:56:07 2008
;;MSG SIZE rcvd: 248
Q2.9 Explain the difference between the Request-URI and the To URI in a SIP INVITE.
50 |
SIP: Understanding the Session Initiation Protocol |
Q2.10 Explain the meaning of each of the parameters in the following Via header field:
Via: SIP/2.0/SCTP room42.lib.edu:4213 ;branch=z9hG4bK3423;received=13.34.3.1
References
[1]Rosenberg, J., H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler, “SIP: Session Initiation Protocol,” RFC 3261, June 2002.
[2]Schulzrinne, H., et al., “RTP: A Transport Protocol for Real-Time Applications,” STD 64, RFC 3550, July 2003.
[3]Handley, M., V. Jacobson, and C. Perkins, “SDP: Session Description Protocol,” RFC 4566, July 2006.
[4]Roach, A., “Session Initiation Protocol (SIP)—Specific Event Notification,” RFC 3265, June 2002.
[5]Campbell, B., et al., “Session Initiation Protocol (SIP) Extension for Instant Messaging,” RFC 3428, December 2002.
[6]Sugano, H., et al., “Presence Information Data Format (PIDF),” RFC 3863, August 2004.
[7]Dierks, T., and C. Allen, “The TLS Protocol Version 1.0,” RFC 2246, January 1999.
[8]Stewart, R., et al., “Stream Control Transmission Protocol,” RFC 2960, October 2000.
[9]Rosenberg, J., H. Schulzrinne, and G. Camarillo, “The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP),” RFC 4168, October 2005.
[10]Rosenberg, J., and H. Schulzrinne, “SIP: Session Initiation Protocol,” RFC 3263, June 2002.
[11]Gulbrandsen, A., P. Vixie, and L. Esibov, “A DNS RR for Specifying the Location of Services (DNS SRV),” RFC2782, February 2000.