1. Session Initiation Protocol (SIP)
Chapter 5

2. Introduction A powerful alternative to H.323 More flexible, simpler
Easier to implement
Advanced features
Better suited to the support of intelligent user devices
A part of IETF multimedia data and control architecture
SDP, RTSP (Real-Time Streaming Protocol), SAP (Session Announcement Protocol)

3. The Popularity of SIP Originally Developed in the MMUSIC
A separate SIP working group
RFC 3261
Many developers
SIP + MGCP/MEGACO
The VoIP signaling in the future
“back-off” or SIPit (SIP Interoperability Tests)
Test products against each other
Organized by SIP Forum

4. The 18th SIPit event in Tokyo, Japan took place April 17-21, 2006, and will be hosted by JPNIC
The 17th SIPit event in Stockholm, Sweden took place to and was hosted by Hotsip
The 16th SIPit event in Banff, Canada took place to and was hosted by Jasomi Networks
The 15th SIPit event in Taiwan took place to and was hosted by CCL/ITRI
The 14th SIPit event in Cannes, France took place to and was hosted by ETSI

5. SIP Architecture A signaling protocol SIP + SDP
The setup, modification, and tear-down of multimedia sessions
SIP + SDP
Describe the session characteristics
Separate signaling and media streams

6. SIP Network Entities Clients Servers
User agent clients
Application programs sending SIP requests
Servers
Responds to clients’ requests
Clients and servers may be in the same platform
Proxy
Acts as both clients and servers

7. Four types of servers Proxy servers
Handle requests or forward requests to other servers
Can be used for call forwarding

8. Redirect servers
Map the destination address to zero or more new addresses
Do not initiate any SIP requests

9. A user agent server A registrar
Accept SIP requests and contacts the user
The user responds → an SIP response
A SIP device
E.g., an SIP-enabled telephone
A registrar
Accepts SIP REGISTER requests
Indicating the user is at a particular address
Typically combined with a proxy or redirect server

10. SIP Call Establishment
It is simple
A number of interim responses

11. SIP Advantages Attempt to keep the signaling as simple as possible
Offer a great deal of flexibility
Various pieces of information can be included within the messages
Including non-standard information
Enable the users to make intelligent decisions
The user has control of call handling
No need to subscribe call features

12. Call Completion to Busy Subscriber service

13. Overview of SIP Messaging Syntax
Text-based
Similar to HTTP
SIP messages
message = start-line
*message-header CRLF
[message-body]
start-line = request-line | status-line
Request-line specifies the type of request
The response line
The success or failure of a given request

14. Message headers Message body
Additional information of the request or response
E.g.,
The originator and recipient
Retry-after header
Subject header
Message body
Describe the type of session
The media format
SDP, Session Description Protocol
Could include an ISDN User Part message
Examined only at the two ends

15. SIP Requests method SP request-URI SP SIP-version CRLF request-URI
The address of the destination
Methods
INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER
extensions: INFO, REFER, UPDATE, …
INVITE
Initiate a session
Information of the calling and called parties
The type of media
～IAM (initial address message) of ISUP
ACK only the final response

16. BYE Options CANCEL Terminate a session
Can be issued by either the calling or called party
Options
Query a server as to its capabilities
A particular type of media
The response if sent an INVITE
CANCEL
Terminate a pending request
E.g., an INVITE did not receive a final response

17. REGISTER INFO Log in and register the address with a SIP server
“all SIP servers” – multicast address ( )
Can register with multiple servers
Can have several registrations with one server
INFO
RFC 2976
Transfer information during an ongoing session
DTMF digits
account balance information
midcall signaling information generated in another network

18. SIP Responses reason-phrase status code
SIP version SP status code SP reason-phrase CRLF
reason-phrase
A textual description of the outcome
Could be presented to the user
status code
A three-digit number
1XX Informational
2XX Success (only code 200 is defined)
3XX Redirection
4XX Request Failure
5XX Server Failure
6XX Global Failure
All responses, except for 1XX, are considered final
Should be ACKed

19. “One number” service

20. SIP Addressing SIP URLs (Uniform Resource Locators) user@host E.g.,
Supplement the URL

21. Message Headers Provide further information about the message E.g.,
～information elements
E.g.,
To:header in an INVITE
The called party
From:header
The caling party
Four main categories
General, request, response, and entity headers
A list in Table 5-2
Mapping in Table 5-3

22. General Headers Used in both requests and responses Basic information
E.g., To:, From:, Call-ID:, …
Contact:
A URL for future communication
May be different from the From: header
Requests passed through proxies

23. Request Headers Response Headers Apply only to SIP requests
Addition information about the request or the client
E.g.,
Subject:
Priority:, urgency of the request
Authorization:, authentication of the request originator
Response Headers
Further information about the response
Unsupported:, features
Retry-After

24. Entity Header Session information presented to the user
Session description, SDP
The RTP payload type, an address and port
Content-Length, the length of the message body
Content-Type, the media type of the message
Content-Encoding, for message compression
Content Disposition,
Content-Language,
Allow, used in a Request to indicate the set of methods supported
Expires, the date and time

25. Example of SIP Message Sequences
Registration
Via:
Call-ID:
host-specific
Content-Length:
Zero, no msg body
Cseg:
Avoid ambiguity
Expires:
TTL
0, unreg
Contact: *

26. Invitation A two-party call Subject: Content-Type: optional
application/sdp

28. Termination of a Call
Cseq:
Has changed

29. Redirect Servers An alternative address Another INVITE
302, Moved temporarily
Another INVITE
Same Call-ID
Cseq ++

30. Proxy Servers Entity headers are omitted Changes the Req-URI Via:
The path
Loop detected, 482
For a response
The 1st Via: header
Checked
removed

32. Proxy state Can be either stateless or stateful Record-Route:
The messages and responses may not pass through the same proxy
Use Contact:
A Proxy might require that it remains in the signaling path
In particular, for a stateful proxy
Insert its address into the Record-Route: header
The response includes the Record-Route: header
The Record-Route: header is used in the subsequent requests
The Route: header = the Record-Route: header in reverse order, excluding the first proxy
Each proxy remove the next from the Route: header

33. Forking Proxy “fork” requests
A user is registered at several locations
;branch=xxx

35. The Session Description Protocol
The message body
SDP, RFC 2327
The Structure of SDP
Session Level Info
Name
The originator
The time
Media Level Info
Media type
Port number
Transport protocol
Media format

36. SDP session description structure

37. SDP Syntax A number of lines of text In each line
field=value
Session-level fields first
Media-level fields
Begin with media description field (m=)

38. Mandatory Fields v=(protocol version)
o=(session origin or creator and session id)
s=(session name), a text string
t=(time of the session)
t=<start time> <stop time>
NTP time values in seconds
m=(media)
m=<media> <port> <transport> <fmt list>
Media type
The transport port
The transport protocol
The media format, an RTP payload format

39. Optional Fileds i=(session information) u=(URI of description)
A text description
At both session and media levels
u=(URI of description)
Where further session information can be obtained
Only at session level
e=( address)
Who is responsible for the session
Only at the session level
p=(phone number)

40. c=(connection information)
Connection type, network type, and connection address
At session or media level
b=(bandwidth information)
In kilobits per second
r=<repeat interval> <active duration> <list of offsets from start- time>
For regularly scheduled session
How often and how many times

41. z=(timezone adjustments)
z=<adjustment time> <offset> <adjustment time> <offset> ....
For regularly scheduled session
Standard time and Daylight Savings Time
k=(encryption key)
k=<method>:<encryption key>
An encryption key or a mechanism to obtain it
At session or media level
a=(attributes)
Describe additional attributes

42. Ordering of Fields Media level Session Level Media description (m)
Protocol version (v)
Origin (o)
Session name (s)
Session information (i)
URI (u)
address (e)
Phone number (p)
Connection info (c)
Bandwidth info (b)
Time description (t)
Repeat info (r)
Time zone adjustments (z)
Encryption key (k)
Attributes (a)
Media level
Media description (m)
Media info (i)
Connection info (c)
Optional if specified at the session level
Bandwidth info (b)
Encryption key (k)
Attributes (a)

43. Subfields
Field = <value of subfield1> <value of subfield2> <value of subfield3> …
Origin (o)
Username, the originator’s login id or “-”
session ID
A unique ID
Make use of NTP timestamp
version, a version number for this particular session
network type
A text string; IN refers to Internet
address type
IP4, IP6
Address, a fully-qualified domain name or the IP address
o=mhandley IN IP

44. Connection Data Media Information
The network and address at which media data are to be received
Network type, address type, connection address
c=IN IP /127
Media Information
Media type
Audio, video, application, data, or control
Port,
Format
List the various types of media
RTP/AVP payload types
m= audio RTP/AVP
G.728, GSM, G.711

45. Attributes Property attribute value attribute rtpmap attribute
a=sendonly
a=recvonly
value attribute
a=orient:landscape
rtpmap attribute
The use of dynamic payload type
a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>].
m=video RTP/AVP 98
a=rtpmap 98 L16/16000/2

46. Usage of SDP with SIP SIP for the establishment of multimedia sessions
SDP – a structured language for describing the sessions
The entity header

47. Negotiation of Media Fig 5-15 If a mismatch
G.728 is selected
If a mismatch
488 or 606
Not Acceptable
A Warning header
INVITE with multiple media streams
Unsupported should also be returned
With a port number of zero

49. Offer/answer

51. OPTIONS Method
Determine the capabilities of a potential called party

52. SIP Extensions and Enhancements
RFC 2543, March 1999
obsoleted by RFCs 3261,3262, 3263, 3265
Will be enhanced considerably before it becomes an Internet standard
183 – session progress (RFC 3261)
Supported: header (RFC 3261)
Require:
Supported:

53. 183 Session-Progress Message
The addition of a new response
Status code 183
To open a one-way media path
From the called party to calling party
convey information about the progress of the call that is not otherwise classified
Tones or announcements
ACM (address complete message) of SS7
For SIP – PSTN – SIP connections
When a temporary media stream is needed
Note that alerting signal can be
Status code 180 (ringing)
The temporary media stream will be terminated
As soon as the called user answers

54. The SIP Supported Header
The Require header
UACs tell UASs about options that the UAC expects the UAS to support
require: 100rel
may receiver 420 (Bad Extension)
The Supported header
enumerates all the extensions supported by the UAC or UAS
Included in both requests and responses
BYE, CANCEL, INVITE, OPTIONS and REGISTER
Should not be included in the ACK
421, extension required
The UAS needs a particular extension to process the request

56. SIP INFO Method A new SIP method – RFC 2976
The transfer of information in the middle of a call
DTMF digits, account-balance information, mid-call signaling information (from PSTN)
A powerful, flexible tool to support new services
e.g., the user’s prepaid account balance

57. SIP Event Notification
SIP-specific event notification
be informed of some event(s)
RFC 3265
SUBSCRIBE
subscribe to certain event
Event: header
NOTIFY
inform the user
200 (OK) response

59. SIP for Instant Messaging
SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions
a working group
RFC 3994, 3856
The exchange of content between a set of participants in near real time
IMs are usually grouped together into brief live comversations
MESSAGE request, RFC 3994
a message body in the form text/plain, or message/cpim (common presence and instant message) using XML

60. Doesn’t establish a SIP dialog
Can be associated with an existing SIP dialog
Contact: header is forbidden
No Record-Route: or Route: header

62. REFER Method RFC 3515 Instruct the receiver to contact a third party
Refer-to:
Can be interpreted as an implicit SUBSCRIBE
The sender will be notified the result
202 (accepted)
An extension
A SIP message is tunneled within a SIP message
Refer-by:

65. Reliability of Provisional Responses
100 (trying), 180 (ringing), 183 (session in progress)
Are not answered with an ACK
If the messages is sent over UDP
Unreliable
Lost provisional response may cause problems when interoperating with other network
180, 183 → Q931 alerting or ISUP ACM
To drive a state machine
E.g., a call to an unassigned number
ACM to create a one-way path

66. RSeq Rack PRACK the option tag Should not Response seq +1, when retxm
Response ACK
PRACK
Prov Resp ACK
the option tag
100rel
Should not
Apply to 100
hop-by-hop

68. UPDATE Method RFC 3311 Change the media format in the early state
re-INVITE cannot be used; it changes the state
Can be used to reserve network resources

69. Integration of SIP and Resource Mang
RFC 3312
The signaling might take a different path from the media
Assume resource-reservation mechanisms available (Chapter 8)
On a per-session basis
On an aggregate basis
A new SIP header in the INVITE
Resources reservation is needed
The user should not yet be alerted
But unrecognized header is ignored

70. Integration of Resource Management and SIP for IP Telephony
A new method, PRECONDITION-MET
The far-end phone will not ring until
Also specifies extensions to SDP
Can define any number of preconditions in SDP without revise SIP every time
The response is sent using reliable signaling
Once the resource is reserved
An UPDATE request is sent
If failed, could select a lower-bandwidth codec

72. The preconditions/requirements in the SDP
Three status
desired, current, and confirmed
Resource reservation
end-to-end (e2e), local, and remote
Purpose
send, recv, and sendrecv
Strength
mandatory, optional, none and failure

73. Examples m=audio 4444 RTP/RTCP 0 a=curr: qos e2e none
a=des: qos mandatory e2e sendrecv
a=curr: qos e2e send
a=curr: qos e2e sendrecv

75. Usage of SIP for Features/Services
Personal mobility by registration
Can carry MIME (Multi-Purpose Internet Mail Extension) content
Text, HTML documents, an image, etc.
SIP address is a URL
Click-to-call applications
Supplementary Custom Local Area Signaling Service (CLASS) services
Call waiting, call forwarding, multi-party calling, call screening
Proxy-controlled: QoS, IN SCP, INAP, OSA

76. Call Forwarding
On busy
486, busy here

77. Consultation Hold
A SIP UPDATE

78. Interworking PSTN Interworking PSTN – SIP – PSTN A SIP URL
A network gateway
Fig. 5-27
SIP to PSTN call
Fig. 5-28
PSTN to SIP call
PSTN – SIP – PSTN
MIME media types
For ISUP and QSIG

81. Interworking with H.323 An Internet draft
SIP-H.323 interworking gateway

86. Summary The future for signaling in VoIP networks Simple, yet flexible
Easier to implement
Fit well with the media gateway control protocols