Professional Documents
Culture Documents
www.iitelecom.com
Audio
G.711 G.723.1 G.726 G.728 G.729
Video
Data
H.261 H.263
T.120
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H.323 terminal
MCU
Network
IP
Gateway
Access Server
PSTN
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PBX
Signaling protocol for multimedia applications Independent of sub layer protocols (TCP, UDP) Standard developed by the IETF (MMUSIC working group) - RFC 2543 SIP works in various phases of the call Localization of the corresponding terminal Analyze recipient profile and resources Negotiation of the media type and of the communication parameters Availability of the correspondent Call set-up and call follow-up SIP uses several existing protocols Message format (HTTP 1.1) Media negotiation (SDP - Session Description Protocol), Media (RTP) Name resolution and mobility (DNS and DHCP) Applications encoding (MIME)
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Audio
G.711 G.729 G.723.1
Signaling
SIP
SDP
RTP/RTCP
TCP/UDP IP Physical
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Registering an IP phone
3. Register 4. 200 OK
Registration Server
1. Register
2. 100 Trying
5. 200 OK
Proxy Server
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SIP Addressing
SIP Addresses are identified by URL, in the form user@host
user = name or telephone number host = domain name or IP addresses
Examples
sip:xyz@iitelecom.com sip:xyz@192.168.10.1 sip:5141234567@iitelecom.com; user=phone
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SIP Components
User Agent
An end user application initiating, receiving and terminating a call
Proxy Server
An application server conveying the requests on behalf of the end user application
The request is processed and sent to the destination (called person) or to another server
Redirect Server
An application server determining the destination address (To:) and returning it to the end user application
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Registration Server
Accept registration requests from the client applications Generally, the service is offered by the Proxy Server or Redirect Server
DNS Server
Used to locate the Proxy Server or Redirect Server
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Registrar
Redirect
Register I am here
Redirect Here is the address Proxy INVITES I will call it for you.
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(User Agent)
session
SDP is similar to H.245 in H.323 functions SDP contains the following parameters:
Example: INVITE
INVITE sip:pierre@192.168.1.31 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v = (protocol version) O = (owner/creator and session to identify) C = (session information) T = (time the session is active) m = (media name and address transport)
Session name
Session duration Contact etc
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SDP Messages
v=0
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Bye
Signaling 200 OK
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Marie 192.168.1.20
Pierre 192.168.1.31
INVITE sip:pierre@192.168.1.31 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v=0 o=marie 3123 121231 IN IP4 192.190.132.20 c=IN IP4 192.190.132.20 m=audio 5004 RTP/AVP 0
INVITE
100 Trying 180 Ringing
SIP/2.0 200 OK Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v=0 o=pierre 5664 456456 IP IP4 192.190.132.31 c=IN IP4 192.190.132.31 m=audio5004 RTP/AVP 0
200 OK
ACK sip:pierre@192.190.132.31 SIP/2.0 Via: SIP/2.0/UDP 192.190.132.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 ACK
ACK
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Use same syntax and semantics as HTTP Request Method (INVITE, ACK, BYE, etc.) Header (Accept, Contact, etc.) Answer Status code (200 OK, 180 Ringing, etc.) Header (Content-type, Content-encoding, etc.)
SIP Methods
INVITE Initiate a call by inviting a user to take part in a session.
SIP Answers
ACK
BYE CANCEL
REGISTER To register the User Agent. OPTIONS Used to know the capacities of the server.
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Location Server
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INVITE pierre@uqam.ca
Pierre?
Pierre@stanford.edu
4 INVITE pierre@stanford.edu
From: marie@iitelecom.com
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Location server
Pierre?
Pierre@stanford.edu
1 From: marie@iitelecom.com
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302 Moved Contact: pierre@stanford.edu
INVITE pierre@uqam.ca
Redirect Server
5 ACK
marie@iitelecom.com
6 INVITE pierre@stanford.edu
From: marie@iitelecom.com
7 200 OK
8 ACK
pierre@stanford.edu
Established session
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ACK
INVITE 180 Ringing 200 OK ACK ACK
180 Ringing
200 OK
Established session
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Established session Bye (also C) Bye (also C) 200 OK INVITE (req A) INVITE (req A) 100 Trying 180 Ringing 180 Ringing 200 OK
200 OK
200 OK
ACK
ACK
Established session
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7. 100 Trying
Proxy Server A
Proxy Server B
SIP Phone A
SIP Phone B
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Proxy Server A
Proxy Server B
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Proxy Server A
Proxy Server B
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Proxy Server A
Proxy Server B
SIP Phone C
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Redirect Server 17. 180 Ringing 16. 180 Ringing 20. 180 Ringing 15. 180 Ringing 19. 180 Ringing
Proxy Server A
Proxy Server B
SIP Phone C
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Redirect Server
25. 200 OK
Proxy Server A
Proxy Server B
SIP Phone C
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SIP Security
Security Mechanisms
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SIP Security
Confidentiality
Integrity
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Registration Hijacking
Authentication
Server Authentication
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Confidentiality
Denial of Service
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User Mobility
Terminal Mobility
Terminal Moving between networks
Session Mobility
User moving between terminals, in runtime
User Mobility
Service Mobility
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Terminal can move between sub networks GSM and WLAN makes it possible Mobile hosts use Register message to inform their server about their
new locations.
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3G and SIP
SIP handles the movement of the mobile user from one domain to
another
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Benefits of 3G
Mobility
Connectivity
Reachability
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SIP References
Columbia university Web site
http://www.cs.columbia.edu/sip/
http://ietf.org/html.charters/sip-charter.html
SIP forum
http://www.sipforum.org
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Questions?
?
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