1. School of Electronics and Information
Introduction to SIP
School of Electronics and Information
Kyung Hee University
Choong Seon HONG

2. Overview of SIP
Proposed standard released in 1999, current release in 2002
Session-layer control-plane signaling protocol with support for establishing, modifying and terminating one-to-one or multiparty sessions.
Light-weight, ASCII based generic signaling protocol to facilitate multimedia communications over IP
Independent of media characteristics and transport protocol properties
Work in IETF currently by SIP WG and SIPPING WG
Related work by MMUSIC WG, AAA WG, GEOPRIV WG, SIMPLE WG, Internet Telephony WG
3GPP has decided to support it in IP multimedia services (IMS) domain.

3. Overview of SIP (cont’d)
SIP is complimentary to SGCP/MGCP
SIP Provides Session Control
SGCP/MGCP Provides Device Control

4. The Big FAQ and SIP? Q: You are too IP-centric, aren’t you?
A: Of course, we are.
Internet telephony (which has Internet in its name) is about IP.
IP telephony runs on top of IP and utilizes the IP service model.
It is not about re-engineering PSTN -- PSTN is good enough.
SIP is much more similar to HTTP rather than to legacy signaling both in terms of service model and protocol design.

5. Problem description
Environment: aging PSTN equipment, carrier fiber overcapacity and arrival of multimedia communications with preparation to millions of mobile hosts that are IP-aware. Additionally, the possibility of programming PSTN-services in a more open environment.
Problem: how to enable gradual deployment and transition to avoid the one-big-leap-for-mankind scenarios?
Approach: IETF multimedia architecture and interoperability schemes with PSTN and 3GPP. Signaling is one part of this. The multimedia architectures capabilities are related to Quality of Service architecture and IP multicast.

6. IETF Multimedia Architecture

7. What Protocols Are Needed?
Signaling protocol to establish presence, locate users, set up, modify and teardown sessions
Media Transport Protocols for transmission of packetized audio/video
Supporting Protocols
Gateway Location, QoS, interdomain AAA,
address translation, IP, etc.
AAA = Authentication, Authorization, Accounting

8. What Protocols Are There
Signaling: SIP/SDP (IETF), H.323 (ITU-T)
Note: SIP adopted by 3gpp; lower production and operation costs reported
Media: RTP (IETF’s, adopted by ITU-T)
Transport: UDP, TCP, (Stream Control Transmission Protocol – RFC 2960)
Supporting protocols:
DNS
TRIP - Telephony Routing over IP - discovery and exchange of IP telephony gateway routing tables between providers
RSVP - Resource Reservation Setup Protocol
COPS - Common Open Policy Service - protocol for for supporting policy control over QoS
Diameter - Authentication, Accounting, Authorization

9. SIP Signaling SIP is end-to-end, client-server session protocol
SIP’s primarily provides presence and mobility
Protocol primitives: Session setup, termination, changes
Arbitrary services built on top of SIP, e.g.:
Redirect calls from unknown callers to secretary
Reply with a webpage if unavailable
Send a JPEG on invitation
Features:
Textual encoding (telnet, tcpdump compatible)
Programmability

10. SIP - General Purpose Presence Protocol
SIP is not limited to Internet
SIP establishes user presence
SIP messages can convey arbitrary signaling payload: session description, instant messages, JPEGs, any MIME types
Suitable for applications having a notion of session
distributed virtual reality systems,
network games (Quake II/III implementations),
video conferencing, etc.
Applications may leverage SIP infrastructure (Call Processing, User Location, Authentication)
Instant Messaging and Presence
SIP for Appliances

11. Internet Multimedia Real Time Protocol (RTP) – media packets
Real Time Control Protocol (RTCP) – monitor & report
Session Announcement Protocol (SAP)
Session Description Protocol (SDP)
Session Initiation Protocol (SIP)
Real Time Stream Protocol (RTSP) – play out control
Synchronized Multimedia Integration Language (SMIL) – mixes audio/video with text and graphics : August 2001
References: Search keyword at
For SMIL -

12. Stages of IP Signaling Development
Precommercial stage ( )
Research in organizations and universities
IETF: Audio/Video Transport (AVT) WG - RTP
IETF: Multiparty Multimedia Session Control (MMUSIC) WG – SIP
PC-centric stage ( )
First commercial VoIP software, proprietary software
Calls from multimedia PC to another multimedia PC
ITU: May-June 1995: H.323v1
Most commercial applications H.323 compliant by the end of 1996
Carrier-grade stage (1998-)
Service provider VoIP deployment
First obstacle: integration to PSTN signalling (gateway functionality)
Development of media gateway controller protocols (MGCP and Megaco/H.248)
Todays approach: coexistence of SIP, H.323, MGCP and Megaco/H.248

13. SIP History Work began in 1995 in IETF mmusic WG
02/1996: draft-ietf-mmusic-sip-00: 15 ASCII pages, one request type
12/1996: ASCII pages, 2 request types
01/1999: ASCII pages, 6 methods
03/1999: RFC 2543, 153 ASCII pages, 6 methods
11/1999: SIP WG formed
11/2000: draft-ietf-sip-rfc2543bis-02, 171 ASCII pages, 6 methods
12/2000: it was recognized that amount of work at SIP WG was becoming unmanageable; 1 RFC; 18 I-Ds on WG’s agenda; numerous individual submissions
04/2001: proposal for splitting SIP WG into SIP and SIPPING announced
2001: SIP implementations widely available

14. SIP End-devices User Agent (user application)
UA Client (originates calls)
UA Server (listens for incoming calls)
both SW and HW available

15. SIP Components SIP Proxy Server SIP Redirect Server SIP Registrar
relays call signaling, i.e. acts as both client and server
operates in a transactional manner, i.e., it keeps no session state
SIP Redirect Server
redirects callers to other servers
SIP Registrar
accept registration requests from users
maintains user’s whereabouts at a Location Server (like GSM HLR)

16. SIP/RTP Media Architecture
Telephony on the Internet may not be a stand-alone business, but part of IP services
SIP/RTP Media Architecture
Public IP Backbone
Goes everywhere
End-to-end control
Consistent for all services
DNS – mobility
Messaging
Web
Directory
Security
QoS
Media services
Sessions
Telephony
…………
Telephone Gateway SIP client
CAS, Q.931, SS7 MG SG
SIP
PCM
RTP
MGCP
CAS : channel associated signaling
Any other sessions

17. Commercial Grade IP Telephony
Assure baseline PSTN features
Leverage and Commonality of telephony with the Web/Internet
New services (new revenue)
Scalability (Web-like)
Baseline PSTN&PBX features
Client & user authentication
Accounting assured QoS
QoS assured signaling
Security assured signaling
Hiding of caller ID & location
Better than PSTN features
New & fast service creation
Internet (rapid) scalability
Mobility
Dynamic user preferences
End-to-end control
Service selection
Feature control
Mid-call control features
Pre-call
Mid-call

18. Complete integration of all services under full user control
IP Communications
Complete integration of all services under full user control
PSTN/PBX-like:
POTS
AIN CS-1, CS-2
PBX & Centrex
User has control of:
All addressable devices
Caller and called party preferences
Better quality than 3.1 kHz
Web-like:
Presence
Voice and text chat
Messaging
Voice, data, video
Multiparty
Conferencing
Education
Games
Any quality
Most yet to be invented

19. Development of SIP IETF - Internet Engineering Task Force
MMUSIC - Multiparty Multimedia Session Control Working Group
SIP developed by Handley, Schulzrinne, Schooler, and Rosenberg
Submitted as Internet-Draft 7/97
Assigned RFC 2543 in 3/99
Internet Multimedia Conferencing Architecture.
Alternative to ITU’s H.323
H.323 used for IP Telephony since 1994
Problems: No new services, addressing, features
Concerns: scalability, extensibility

20. SIP Philosophy Internet Standard
IETF -
Reuse Internet addressing (URLs, DNS, proxies)
Utilizes rich Internet feature set
Reuse HTTP coding
Text based
Makes no assumptions about underlying protocol:
TCP, UDP, X.25, frame, ATM, etc.
Support of multicast

21. SIP Clients and Servers - 1
SIP uses client/server architecture
Elements:
SIP User Agents (SIP Phones)
SIP Servers (Proxy or Redirect - used to locate SIP users or to forward messages.)
Can be stateless or stateful
SIP Gateways:
To PSTN for telephony interworking
To H.323 for IP Telephony interworking
Client - originates message
Server - responds to or forwards message

22. SIP Clients and Servers - 2
Logical SIP entities are:
User Agents
User Agent Client (UAC): Initiates SIP requests
User Agent Server (UAS): Returns SIP responses
Network Servers
Registrar: Accepts REGISTER requests from clients
Proxy: Decides next hop and forwards request
Redirect: Sends address of next hop back to client
The different network server types may be collocated

23. SIP Addressing SIP gives you a globally reachable address
Callees bind to this address using SIP REGISTER method.
Callers use this address to establish real-time communication with callees.
URLs used as address data format; examples:
geo.position:=48.54_ _120

24. SIP Addressing (cont’d)
must include host, may include user name, port number, parameters (e.g., transport), etc.
may be embedded in Webpages, signatures, printed on your business card, etc.
address space unlimited
non-SIP URLs can be used as well (mailto:, ...)

25. SIP Registration

26. SIP Session Setup Example
User Agent
Client
SIP
User Agent
Server
INVITE
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
sip.uunet.com

27. Proxy Server Example 200 OK BYE host.wcom.com ACK sip.uunet.com SIP
INVITE
host.wcom.com
ACK
INVITE
sip.uunet.com
SIP
User Agent
Client
Proxy
Server
Media Stream
server.wcom.com

28. SIP Operation in Proxy Mode

29. Proxy Server Functionality
Serve as rendezvous point at which callees are glabally reachable
Perform routing function, i.e., determine to which hop (UA/proxy/redirect) signaling should be relayed
Allow the routing function to be programmable. Arbitrary logic may be built on top of the protocol
user’s signaling preferences
AAA
firewall control
etc.
Forking: Several destinations may be tried for a request sequentially or in parallel.

30. Proxy Chaining
There may be also cases when a local outbound proxy may be involved
provides locally important call processing logic (e.g., identifying nearest 119)
manages firewall
provides least-gateway-cost routing service
IP phones must know address of the proxy:may be configured manually or with a configuration protocol (DHCP, TFTP, ... )
In general, servers may be arbitrarily chained
a central company’s server may distribute signaling to departmental servers
a user may want to forward incoming calls to her cell phone

31. Proxy Chaining – an Example
Note : signaling (in red) may take a completely different path from media in blue)

32. Redirect Server Example
302 Moved
ACK
Media Stream
INVITE
SIP
User Agent
Client
Redirect
Server
180 Ringing
INVITE
REGISTER
host.wcom.com
sip.uunet.com
200 OK
server.wcom.com C RS
UAS 1 2 3

33. SIP Operation in Redirect Mode

34. SIP Server - Proxy versus Redirection
A SIP server may either proxy or redirect a request
Which of the two method applies is a configuration issue. It may be statically configured or dynamically determined/
Redirection useful if a user moves or changes her provider (PSTN: “The number you have dialed is not available.”) -- caller does not need to try the original server next time. Stateless.
Proxy useful if forking, AAA, firewall control needed. In general, proxying grants more control to the server.

35. SIP Requests : RFC2543 Methods
SIP Requests (Messages) defined as:
Method SP Request-URI SP SIP-Version CRLF (SP=Space, CRLF=Carriage Return and Line Feed)
Example: INVITE SIP/2.0

36. } SIP Requests Example Required Headers (fields):
Via: Shows route taken by request.
Call-ID: unique identifier generated by client.
CSeq: Command Sequence number
generated by client
Incremented for each successive request
INVITE SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston
To: Jean Luc Picard
Call-ID:
CSeq: 1 INVITE }
Uniquely identify this session request

37. SIP Requests Example Typical SIP Request:
INVITE SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston
To: Jean Luc Picard
Call-ID:
CSeq: 1 INVITE
Contact:
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 124
v=0
o=ajohnston IN IP4 host.wcom.com
s=Let's Talk
t=0 0
c=IN IP
m=audio RTP/AVP 0 3

38. SIP Responses SIP Responses defined as (HTTP-style):
SIP-Version SP Status-Code SP Reason-Phrase CRLF (SP=Space, CRLF=Carriage Return and Line Feed)
Example: SIP/ Not Found
First digit gives Class of response:

39. SIP Responses Example Required Headers:
Via, From, To, Call-ID, and CSeq are copied exactly from Request.
To and From are NOT swapped!
SIP/ OK
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston
To: Jean Luc Picard
Call-ID:
CSeq: 1 INVITE

40. SIP Responses Example Typical SIP Response (containing SDP)
SIP/ OK
Via: SIP/2.0/UDP host.wcom.com
From: Alan Johnston
To: Jean Luc Picard
Call-ID:
CSeq: 1 INVITE
Contact:
Subject: Where are you these days?
Content-Type: application/sdp
Content-Length: 107
v=0
o=picard IN IP4 uunet.com
s=Engage!
t=0 0
c=IN IP
m=audio 3456 RTP/AVP 0

41. Forking Proxy Example Fork S1 C S2 SIP User Agent Client SIP Proxy
Server
SIP
User Agent
Server 1
SIP
User Agent
Server 2
INVITE S1
INVITE
INVITE C
100 Trying
404 Not Found S2
ACK
Fork
180 Ringing
180 Ringing
200 OK
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
proxy.wcom.com
sip.mci.com
sip.uunet.com

42. SIP Headers - Partial List

43. SIP Headers - Continued

44. SIP Headers - Continued

45. Via Headers and Routing
Via headers are used for routing SIP messages
Requests
Request initiator puts address in Via header
Servers check Via with sender’s address, then add own address, then forward. (if different, add “received” parameter)
Responses
Response initiator copies request Via headers.
Servers check Via with own address, then forward to next Via address

46. SIP Firewall Considerations
Firewall Problem
Can block SIP packets
Can change IP addresses of packets
TCP can be used instead of UDP
Record-Route can be used:
ensures Firewall proxy stays in path
A Firewall proxy adds Record-Route header
Clients and Servers copy Record-Route and put in Route header for all messages

47. SIP Message Body Message body can be any protocol
Most implementations:
SDP - Session Description Protocol
RFC /98 by Handley and Jacobson
Used to specify info about a multi-media session.
SDP fields have a required order
For RTP - Real Time Protocol Sessions:
RTP Audio/Video Profile (RTP/AVP) payload descriptions are often used

48. Session Description Protocol (SDP)
Convey sufficient information to enable participation in a multimedia session
SDP includes description of:
Media to use (codec, sampling rate)
Media destination (IP address and port number)
Session name and purpose
Times the session is active
Contact information
Note: indeed SDP is a data format rather than a protocol.

49. SDP Examples
v=0
o=sisalem IN IP
s=SIP Tutorial
c=IN IP t=
m=audio RTP/AVP 0 98
a=rtpmap:98 L16/11025/2

50. SDP Examples (cont’d) SDP Example SDP Example v=0
o=ajohnston IN IP4 host.wcom.com
s=Let's Talk
t=0 0
c=IN IP
m=audio RTP/AVP 0 3
SDP Example
v=0
o=picard IN IP4 uunet.com
s=Engage!
t=0 0
c=IN IP
m=audio 3456 RTP/AVP 0

51. Another SDP Example (cont’d)
v=0
o=alan IN host.wcom.com
s=SSE University Seminar - SIP
i=Audio, Listen only u= p=
c=IN IP
b=CT:128 t=
m=audio 3456 RTP/AVP 0 3
a=type:recvonly

52. Authentication & Encryption
SIP supports a variety of approaches:
end to end encryption
hop by hop encryption
Proxies can require authentication:
Responds to INVITEs with 407 Proxy-Authentication Required
Client re-INVITEs with Proxy-Authorization header.
SIP Users can require authentication:
Responds to INVITEs with 401 Unathorized
Client re-INVITEs with Authorization header

53. SIP Encryption Example
INVITE SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
From: Alan Johnston
To: Jean Luc Picard
Call-ID:
CSeq: 1 INVITE
Content-Length: 224
Encryption: PGP version=2.6.2, encoding=ascii
hWjasGdg,ddgg+fdgf_ggEO;ALewAKFeJqAFSeDlkjhasdfkj!aJsdfasdfKlfghgasdfasdfa|Gsdf>a!sdasdf3w29451k45mser?we5y;343.4kfj2ui2S8~&djGO4kP%Hk#(Khujefjnjmbm.sd;da’l;12’;123=]aw;erwAo3529ofgk

54. PSTN Features with SIP Features implemented by SIP Phone
Call answering: 200 OK sent
Busy: 483 Busy Here sent
Call rejection: 603 Declined sent
Caller-ID: present in From header
Hold: a re-INVITE is issued with IP Addr =
Selective Call Acceptance: using From, Priority, and Subject headers
Camp On: 181 Call Queued responses are monitored until 200 OK is sent by the called party
Call Waiting: Receiving alerts during a call

55. PSTN Features with SIP Features implemented by SIP Server
Call Forwarding: server issues 301 Moved Permanently or 302 Moved Temporarily response with Contact info
Forward Don’t Answer: server issues 408 Request Timeout response
Voic server 302 Moved Temporarily response with Contact of Voic Server
Follow Me Service: Use forking proxy to try multiple locations at the same time
Caller-ID blocking - Privacy: Server encrypts From information

56. SIP User Location Example
SIP supports mobility across networks and devices
Q=quality gives preference
SIP/ Moved temporarily
Contact:
;service=IP,voice mail
;media=audio ;duplex=full ;q=0.7
Contact : phone: ; service=ISDN
;mobility=fixed; language=en,es, ;q=0.5
Contact : phone: ; service=pager
;mobility=mobile
;duplex=send-only ;media=text; q=0.1; priority=urgent
;description=“For emergency only”
Contact : mailto:

57. Programming SIP Examples Users and third parties may program
“discard all calls from Monica during my business hours”
“redirect authenticated friends to my cell phone, anyone else to my secretary”
“if busy, return my homepage and redirect to recorder”
Users and third parties may program
SIP follows HTTP programming model
Mechanisms suggested in IETF: CGI, Call Processing Language (CPL), Servlets

58. Call Processing Logic Example

59. SIP Mobility Support 4 Mobile Host SIP Redirect Server 5 SIP Proxy
Foreign Network
Home
Network 3 1 2 6 7
1 INVITE
moved temporarily
3, 4 INVITE
5, 6 OK
7 Data
Corresponding
Host
Global: Wire and wireless
No tunneling required
No change to routing
For fast hand-offs use:
Use Cellular IP or
Use DRCP
* Dynamic Registration and Configuration Protocol

60. Need not bother home registrar: Use multi-stage registration
SIP Mobility
Pre-call mobility
MH can find SIP server via multicast REGISTER
MH acquires IP address via DHCP
MH updates home SIP server
Mid-call mobility
MH->CH: New INVITE with Contact and updated SDP
Re-registers with home registrar
Need not bother home registrar: Use multi-stage registration
Recovery from disconnects

61. Mobile IP Communications
Mobile IP Requirements
Transparency above L2:
Move but keep IP address and all sessions alive
Mobility
Within subnet
Within domain
Global
AAA and NAIs
Location privacy
QoS for r.t. communications
Evolution of Wireless Mobility
Circuit Switched Mobility
based on central INs
LAN-MAN: Cellular IP
Wide Area: Mobile IP
Universal (any net): SIP

62. Presence, Instant Messaging and Voice

63. IP SIP Phones and Adaptors
Are Internet hosts
Choice of application
Choice of server
IP appliance
Implementations
3Com (2)
Cisco
Columbia University
Mediatrix (1)
Nortel (3)
Pingtel 1
                 2 3

64. H.323/SIP Comparison

65. H.323 vs. SIP: Basic Call Control

66. H.323 vs. SIP: Advanced features

67. H.323 vs. SIP Scalability

68. H.323 vs. SIP Extensibility of functionality

69. H.323 vs. SIP Ease of customization

70. H.323 vs. SIP Ease of customization

71. H.323 vs. SIP Ease of Implementation

72. Summary :SIP vs H.323

73. Internet Drafts
Session Timers in the Session Initiation Protocol (SIP)
Caller Preferences for the Session Initiation Protocol (SIP)
Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP)
The Stream Control Transmission Protocol as a Transport for for the Session Initiation Protocol
The Session Inititation Protocol (SIP) 'Replaces' Header
The SIP Referred-By Mechanism
Compressing the Session Initiation Protocol
Session Initiation Protocol Extension to Assure Congestion Safety
A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages
The Session Inititation Protocol (SIP) 'Join' Header
SIP Authenticated Identity Body (AIB) Format
S/MIME AES Requirement for SIP
An Extension to the Session Initiation Protocol for Request History Information
Communications Resource Priority for the Session Initiation Protocol (SIP) Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)
Connection Reuse in the Session Initiation Protocol (SIP)
The Internet Assigned Number Authority Universal Resource Identifier Parameter Registry for the Session Initiation Protocol
The Internet Assigned Number Authority Header Field Parameter Registry for the Session Initiation Protocol
Session Initiation Protocol (SIP) Extension for Event State Publication
Interactions of Preconditions with Session Mobility in the Session Initiation Protocol (SIP)
Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)

74. RFCs The SIP INFO Method (RFC 2976)
MIME media types for ISUP and QSIG Objects (RFC 3204)
SIP: Session Initiation Protocol (RFC 3261)
Reliability of Provisional Responses in SIP (RFC 3262)
SIP: Locating SIP Servers (RFC 3263)
SIP-Specific Event Notification (RFC 3265)
DHCP Option for SIP Servers (RFC 3361)
Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA) (RFC 3310)
The Session Initiation Protocol UPDATE Method (RFC 3311)
Integration of Resource Management and SIP (RFC 3312)
Internet Media Type message/sipfrag (RFC 3420)
A Privacy Mechanism for the Session Initiation Protocol (SIP) (RFC 3323)
Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks (RFC 3325)
Session Initiation Protocol Extension for Instant Messaging (RFC 3428)
The Reason Header Field for the Session Initiation Protocol (SIP) (RFC 3326) Session Initiation Protocol Extension for Registering Non-Adjacent Contacts (RFC 3327)
Security Mechanism Agreement for the Session Initiation Protocol (SIP) Sessions (RFC 3329)
Private Session Initiation Protocol (SIP)Extensions for Media Authorization (RFC 3313) The Session Initiation Protocol (SIP) Refer Method (RFC 3515)
Dynamic Host Configuration Protocol (DHCPv6)Options for Session Initiation Protocol (SIP) Servers (RFC 3319)
An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing (RFC 3581)
Session Initiation Protocol Extension Header Field for Service Route Discovery During Registration (RFC 3608)

75. Relevant IETF Working Groups
Audio/Video Transport (avt) - RTP
Differentiated Services (diffserv) – QoS in backbone
IP Telephony (iptel) – CPL, GW location, TRIP
Integrated Services (intserv) – end-to-end QoS
Media Gateway Control (megaco) – IP telephony gateways
Multiparty Multimedia Session Control (mmusic) – SIP, SDP, conferencing
PSTN and Internet Internetworking (pint) – mixt services
Resource Reservation Setup Protocol (rsvp)
Service in the PSTN/IN Requesting InTernet Service (spirits)
Session Initiation Protocol (sip) – signaling for call setup
Signaling Transport (sigtran) – PSTN signaling over IP
Telephone Number Mapping (enum) – surprises !
Instant Messaging and Presence Protocol (impp)

76. SIP Summary SIP is: SIP is not: Relatively easy to implement
Gaining vendor and carrier acceptance
Very flexible in service creation
Extensible and scaleable
Appearing in products right now
SIP is not:
Going to make PSTN interworking easy
Going to solve all IP Telephony issues (QoS)

77. References
Book on “Internetworking Multimedia” by Jon Crowcroft, Mark Handley, Ian Wakeman, UCL Press, 1999 by Morgan Kaufman (USA) and Taylor Francis (UK)
RFC 3261: “SIP: Session Initiation Protocol”
The IETF SIP Working Group home page
SIP Home Page
Papers on IP Telephony