1. RFC 3261+ Session Initiation Protocol (SIP)
AN Lab
May 19, 2006
Kim, YangJung

2. Presentation Outline ₪ PART I ₪ ₪ PART II ₪ What is SIP?
SIP Protocol Updates
SIP Protocol Structure
SIP Users / Work groups
₪ PART II ₪
SIP in a similar Domain
SIP in a dissimilar Domain
Proxy servers in SIP
Complimentary protocols in SIP
Concerns about SIP
SIP a new generation of service
Advanced Network Laboratory Kim, YangJung

3. What is SIP?
SIP (Session Initiation Protocol) is an application-layer control protocol that can establish, modify and terminate multimedia sessions such as Internet telephony calls (VOIP), multimedia distribution, multimedia conferences, chat, interactive games and virtual reality.
SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body.
SIP is a text-based protocol that uses UTF-8 encoding
SIP uses port 5060 both for UDP and TCP. SIP may use other transports
SIP does not provide services. Rather, SIP provides primitives that can be used to implement different services.
Advanced Network Laboratory Kim, YangJung

4. What is SIP? (cont.)
SIP supports 5 facets of establishing and terminating multimedia communications:
User location: determination of the end system to be used for communication
User availability: determination of the willingness of the called party to engage in communications
User capabilities: determination of the media and media parameters to be used
Session setup: "ringing", establishment of session parameters at both called and calling party
Session management: including transfer and termination of sessions, modifying session parameters, and invoking services
Advanced Network Laboratory Kim, YangJung

5. SIP Protocol updates
1996
Mark Hadley’s SIP(Session Invitation Protocol)
Henning Schulzrinne’s SCIP(Simple Conference Control Protocol)
RFC 2543 by IETF MMUSIC WG
IETF SIP WG
RFC 2543bis
RFC 3261
Advanced Network Laboratory Kim, YangJung

6. SIP Protocol updates
RFC2543: Session Initiation Protocol [Obsolete] It is also an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these.
RFC3261: Session Initiation Protocol (SIP)
RFC3262: SIP Reliable provisional response messaging: It is an extension to the SIP providing reliable provisional response messages. This extension uses the option tag 100rel and defines the Provisional Response ACKnowledgement (PRACK) method.
RFC3263: Locating SIP server: It uses DNS procedures to allow a client to resolve a SIP Uniform Resource Identifier (URI) into the IP address, port, and transport protocol of the next hop to contact. It also uses DNS to allow a server to send a response to a backup client if the primary client has failed.
Advanced Network Laboratory Kim, YangJung

7. SIP Protocol updates (cont.)
RFC3264: An Offer/Answer Model with the Session Description Protocol (SDP): In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP).
RFC3265: Session Initiation Protocol (SIP)-Specific Event Notification: It is an extension to the SIP. The purpose of this extension is to provide an extensible framework by which SIP nodes can request notification from remote nodes indicating that certain events have occurred.
We need to note that RFC3265 is NOT intended to be a general-purpose infrastructure for all classes of event subscription and notification.
Advanced Network Laboratory Kim, YangJung

8. SIP Protocol structure
SIP is structured as a layered protocol.
Each layer is independent but loosely coupled
Lower layer is encoded in BNF
Transport layer defines how client and server send and receives responses
Transaction layer handles application-layer retransmissions, matching of responses to requests, and application-layer timeouts.
Transaction users comprise all SIP entities except stateless proxies.
Transaction User (TU)
Transaction
Transport
Syntax and Encoding
Advanced Network Laboratory Kim, YangJung

9. SIP Protocol structure
SIP protocol defines several methods:
Methods in SIP RFC
INVITE: When a user agent client desires to initiate a session (for example, audio, video, or a game), it formulates an INVITE request and sends it to one or more user agent server (UAS).
re-INVITE: An INVITE request sent within an existing dialog is known as a re-INVITE to changing addresses or ports, adding a media stream, deleting a media stream, and so on.
REGISTER: Registration entails sending a REGISTER request to a special type of UAS known as a Registrar/Location server.
ACK: Used to facilitate reliable message exchange for INVITEs.
CANCEL: Used to cancel an invitation.
BYE: The BYE request is used to terminate a specific session or attempted session.
OPTIONS: The SIP method OPTIONS allows a UA to query another UA or a proxy server as to its capabilities. This allows a client to discover information about the supported methods, content types, extensions, codecs, etc. without "ringing" the other party.
Advanced Network Laboratory Kim, YangJung

10. SIP Protocol structure (cont.)
Methods extensions from other RFC’s
SIP method info: Extension in RFC 2976
SIP method notify: Extension in RFC 2848 PINT
SIP method subscribe: Extension in RFC 2848 PINT
SIP method unsubscribe: Extension in RFC 2848 PINT
SIP method update: Extension in RFC 3311
SIP method message: Extension in RFC 3428
SIP method refer: Extension in RFC 3515
SIP method prack: Extension in RFC 3262
SIP Specific Event Notification: Extension in RFC 3265
SIP Message Waiting Indication: Extension in RFC 3842
SIP method PUBLISH: Extension is RFC 3903
Advanced Network Laboratory Kim, YangJung

11. SIP Protocol structure (cont.)
SIP responses codes
1xx: Provisional — request received, continuing to process the request;
2xx: Success — the action was successfully received, understood, and accepted;
3xx: Redirection — further action needs to be taken in order to complete the request;
4xx: Client Error — the request contains bad syntax or cannot be fulfilled at this server;
5xx: Server Error — the server failed to fulfill an apparently valid request;
6xx: Global Failure — the request cannot be fulfilled at any server.
Advanced Network Laboratory Kim, YangJung

12. SIP Protocol structure (cont.)
SIP responses example with SDP
Advanced Network Laboratory Kim, YangJung

13. (UAC:User Agent Client)
SIP Architecture
Fundamental Message Procedures
Request
SIP Redirect
Server
Response
Location Service 2
Location
Server 3 5 4 6 1 11 11 7 12 10
SIP Proxy 8
SIP Client
(UAC:User Agent Client) 9
SIP Proxy
SIP Client
(User Agent Server)
Advanced Network Laboratory Kim, YangJung

14. SIP Architecture (E-to-E)
End-to-End Scenario
Advanced Network Laboratory Kim, YangJung

15. SIP Users / Workgroups IP Telephony (VoIP and beyond)
AVT - Audio/Video Transport (RTP)
IPTEL- IP Telephony (CPL, GW location)
SIP- signaling for call setup
MMUSIC– Multiparty Multimedia Session Control
(SIP, SDP, conferencing)
SIPPING- Session Initiation Proposal Investigation
SIP
Vonage
Advanced Network Laboratory Kim, YangJung

16. SIP Users / Workgroups (cont.)
Instant Messaging and Presence
IMPP - Instant Messaging and Presence Protocol
SIMPLE - SIP for Instant Messaging and Presence
Leveraging Extensions
XMPP - open, XML-based protocol for near real-time
extensible messaging and presence
SIP typically is used over UDP or TCP,
it could, without technical changes, be run
over IPX, or carrier pigeons, frame relay,
ATM AAL5 or X.25, in rough order of
desirability.
Advanced Network Laboratory Kim, YangJung

17. SIP
₪ PART II ₪
Advanced Network Laboratory Kim, YangJung

18. SIP Operation Advanced Network Laboratory 18 Kim, YangJung Alice’s
softphone
atlanta.com
proxy
biloxy.com
Bob’s
SIP Phone
100 F3
INVITE F1
180 F7
INVITE F2
100 F5
INVITE F4
180 F6
180 F8
200 OK F9
200 OK F10
200 OK F11
ACK F12
Media Session
BYE F13
200 OK F14
Advanced Network Laboratory Kim, YangJung

19. SIP in a similar Domain 3 4 5 2 6 7 1 1. Call USER B
2. Query “Where is USER B?”
3. Response “USER B SIP address”
4. ‘Proxied’ Call
5. Response
6. Response
7. Multimedia Channel established
Advanced Network Laboratory Kim, YangJung

20. Advanced Network Laboratory 20 Kim, YangJung
SIP in a dissimilar Domain
DOMAIN A.COM 1 10 4 2 11 9 3 8 7 6 5
DOMAIN B.COM
Advanced Network Laboratory Kim, YangJung

21. SIP in a dissimilar Domain
Call USER B
2. Query “How to get to USER B, DOMAIN B?”
3. Response “Address of Proxy controller for Domain.”
4. Call “Proxied” to SIP Proxy for Domain B.
5. Query “Where is USER B?”
6. USER B’s address
7. Proxied call
8. Response
9. Response
10. Response
11. Multimedia Channel established
Advanced Network Laboratory Kim, YangJung

22. Proxy Servers in SIP
SIP is a HTTP-like, textual, client server protocol, using – like address.
Proxy servers take care of setting up sessions betweens users.
Signals and media takes different path
DNS SRV Query? iptel.org
Reply: IP Address of iptel.org SIP server
INVITE:
From:
To: sip:
Call-ID:
INVITE:
From:
To: sip:
Call-ID:
OK 200
From:
To: sip: tag 34
Call-ID:
OK 200
From:
To: sip: tag 34
Call-ID:
Media Stream
Advanced Network Laboratory Kim, YangJung

23. Proxy Servers in SIP (cont.)
Proxy servers maintain central role in SIP network.
They glue SIP components such as phones, gateways, applications and other domains.
They provide place for service implementation (missed calls, forwarding, screening, etc.) and service access control
SMS Gateway
PSTN Gateway
Applications
IP Phone pool
Other Domains
Advanced Network Laboratory Kim, YangJung

24. Complimentary protocols in SIP
SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build a complete multimedia architecture. Typically, these architectures will include protocols are:
Real-time Transport Protocol (RTP) (RFC 1889): for transporting real-time data and providing QoS feedback.
Real-Time streaming protocol (RTSP) (RFC 2326): for controlling delivery of streaming media.
Media Gateway Control Protocol (MEGACO) (RFC 3015): for controlling gateways to the Public Switched Telephone Network (PSTN) and
Session Description Protocol (SDP) (RFC 2327): for describing multimedia sessions.
Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.
Advanced Network Laboratory Kim, YangJung

25. Comparison SIP and H.323 SIP H.323 speed High (simplicity , use UDP)
Low (complexity, use TCP)
multicast
Yes No
URL usage
URL itself
(H.225)URL in H.323protocol
Call prioritization
The priority header field
Overlook
Encoding
Text encoding
Binary encoding(ASN.1)
Advanced Network Laboratory Kim, YangJung

26. Concerns about SIP
NAT Traversal: SIP will make networks more vulnerable, the source may be the firewall and NAT (Network Address Translator) issue. SIP communication doesn't traverse most installed firewalls and NATs.
Built-in delay: VOIP/SIP by it's nature has some built in delay. It digitizes small blocks of your conversation, compresses it, sends it to a PBX in packet form which intern converts it from one format to another if needed, then sends the packets on to their destination which decompress and converts it back into voice.
Advanced Network Laboratory Kim, YangJung

27. SIP a new generation of service
Virtually eliminates long distance cost
More features available in VoIP than conventional phones.
For businesses, VoIP is a great way to manage operating costs in the office giving you economical, reliable, scalable, non pro priority phone system options.
SIP makes it easier to develop and debug applications leading to lower product costs for equipment providers.
SIP a new generation of service to the future .!
Advanced Network Laboratory Kim, YangJung

28. References Iptel.org http://www.iptel.org
RFC search engine
SIP
SIP Knowledge
VoIP Info.
Advanced Network Laboratory Kim, YangJung

29. ₪ Any Questions? ₪ ₪ Any Questions? ₪ Q & A Session
Advanced Network Laboratory Kim, YangJung