SIP TRUNKING
SIP Education
SIP Terminology
TERMS
SIP: Session Initiation Protocol (RFC 3261) – Application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.
SIP Methods: SIP protocol commands or messages (eg: INVITE, BYE)
SIP Response Codes: Responses to SIP Methods indicating success, failure or other information. (eg: 200-Ok)
SIP User Agent (UA): An endpoint device that can issue or respond to SIP protocol methods.
SIP User Agent Client (UAS): A SIP endpoint device issuing the request (eg: Phone,
PC, PDA…).SIP Gateway: A network element that can convert SIP methods and response codes to
another protocol.SIP Proxy Server: An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients.
SDP: Session Description Protocol (RFC 2327): – Text-based protocol describing multi-
media sessions.Softswitch: Software application that coordinates VoIP call switching between endpoints, commonly duplicating
SIP METHODS
REGISTER: Registers a user with a Proxy/Registrar
INVITE: Session setup request or media negotiation. Used also to hold & retrieve calls
CANCEL: Used to cancel an INVITE transaction
ACK: Acknowledgement for an INVITE transaction completion
BYE: Termination a session
OPTIONS: Used as a query for remote’s status & capabilities
INFO: Mid-call signaling information exchange
SUBSCRIBE: Request notification of call events
NOTIFY: Event notification after an explicit/implicit subscription
REFER: Call Transfer request
SIP RESPONSE CODES
100: Trying – Request has been received by a proxy/gateway
180: Ringing – the called party received the INVITE request, the phone is ringing.
181: Call is being forwarded
182: Queued – Invite has been received and will be processed in a queue
183: Session Progress – Used to convey report of incoming early-media
200: OK – successful transaction completion
302: Moved Temporarily – Forwarded call to given contact
305: Use Proxy – Repeat same call setup using a given proxy
400: Bad Request – General error
401: Unauthorized – The server requires client authentication
404: Not Found – The user does not exist at the specified domain
408: Request Timeout
486: Busy here
5xx: Server Failure
6xx: Global Failure
SIP FIELDS
Field |
Meaning |
INVITE Header |
Inviting use at Sip address Bob@Broadway.com to a media session. |
Via |
The response to the INVITE message should be returned to the specified address, using the specified protocol (UDP). |
From |
The calling party. |
To |
The called party. |
Call-ID |
A unique field, used to identify the call. |
Max-Forwards |
The Max-Forwards value is an integer in the range 0-255 indicating the remaining number of times this request message us allowed to be forwarded. |
CSeq |
Command Sequence header. |
Contact |
The SIP Address of the calling party. |
Content-Type |
SIP messages carry bodies that are transparent to the SIP Protocol. |
Content-Length |
The length of the body (in bytes). |
SDP |
Session Description Protocol. |
SIP CALL EXAMPLE
INVITE sip:+17033401234@99.88.77.66:5060 SIP/2.0
Via: SIP/2.0/UDP 66.77.88.99:5060;branch=z9hG4bK506071629343-1157012389004
From: <sip:+16024567890@66.77.88.99>;tag=VPSF506071629343
To: <sip:+17033401234@99.88.77.66:5060>
Call-ID: MCLMGC0120061120200030043630@209.244.63.24
CSeq: 1 INVITE
Contact: <sip:+16024567890@66.77.88.99:5060;transport=udp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 119
Remote-Party-ID: <sip:+16024567890@66.77.88.99;party=calling;screen=no;privacy=off
v=0
o=- 1164052830 1164052831 IN IP4 88.77.66.66
s=-
c=IN IP4 88.77.66.66
t=0 0
m=audio 61162 RTP/AVP 0 18