Session Initiation Protocol (SIP)

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over IP. The protocol can be used for creating, modifying and terminating two-party or multiparty sessions. Sessions may consist of one or several media streams.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on TCP, UDP and SCTP. It is a text-based protocol, incorporating many elements of the HTTP and SMTP.

Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, and presence information and file transfer. The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information. MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer.

Network elements

SIP also defines server network elements. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peer-to-peer, this approach is often impractical for a public service.

User Agent

A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response.

Proxy server

An intermediary entity that acts as both a server (UAS) and a client (UAC) for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it.

Registrar

A server that accepts register requests and places the information it receives in those requests into the location service for the domain it handles which registers one or more IP addresses to a certain SIP URI, indicated by the sip: scheme, although other protocol schemes are possible (such as tel:). More than one user agent can register at the same URI, with the result that all registered user agents will receive a call to the SIP URI.

Redirect server

A user agent server that generates 3xx (Redirection) responses to requests it receives, directing the client to contact an alternate set of URIs. The redirect server allows proxy servers to direct SIP session invitations to external domains.

SIP messages

SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP messages: requests and responses. The first line of a request has a method, defining the nature of the request, and a Request-URI, indicating where the request should be sent. The first line of a response has a response code.

 List of SIP request methods

• REGISTER: Used by a UA to indicate its current IP address and the URLs for which it would like to receive calls.
• INVITE: Used to establish a media session between user agents.
• ACK: Confirms reliable message exchanges.
• CANCEL: Terminates a pending request.
• BYE: Terminates a session between two users in a conference.

·         OPTIONS: Requests information about the capabilities of a caller, without setting up a

List of SIP response codes

• Provisional (1xx): Request received and being processed. Like 100 for Trying, and 180 for Alerting.
• Success (2xx): The action was successfully received, understood, and accepted. Like 200 for OK, and 202 for Accepted.
• Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request. Like 301 for Moved Permanently and 302 for Moved Temporarily
• Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the server. Like  404 User Not Found and 480 Temporarily Unavailable
• Server Error (5xx): The server failed to fulfill an apparently valid request. Like 500 for Server Internal Error or 503 for Service Unavailable
• Global Failure (6xx): The request cannot be fulfilled at any server. Like 603 when the call is declined

 

Class of Response	Status Code	Explanation
Informational/provisional	100	Trying
	180	Ringing
	181	Call Is Being Forwarded
	182	Queued
	183	Session Progress
Success	200	OK
Redirection	300	Multiple Choices
	301	Moved Permanently
	302	Moved Temporarily
	305	Use Proxy
	380	Alternative Service
Client-error	400	Bad Request
	401	Unauthorized
	402	Payment Required
	403	Forbidden
	404	Not Found
	405	Method Not Allowed
	406	Not Acceptable
	407	Proxy Authentication Required
	408	Request Timeout
	410	Gone
	413	Request Entity Too Large
	414	Requested URL Too Large
	415	Unsupported Media Type
	416	Unsupported URI1 Scheme
	420	Bad Extension
	421	Extension Required
	423	Interval Too Brief
	480	Temporarily Not Available
	481	Call Leg or Transaction Does Not Exist
	482	Loop Detected
	483	Too Many Hops
	484	Address Incomplete
	485	Ambiguous
	486	Busy Here
	487	Request Terminated
	488	Not Acceptable Here
	491	Request Pending
	493	Undecipherable
Server-error	500	Internal Server Error
	501	Not Implemented
	502	Bad Gateway
	503	Service Unavailable
	504	Server Timeout
	505	SIP Version Not Supported
	513	Message Too Large
Global failure	600	Busy Everywhere
	603	Decline
	604	Does Not Exist Anywhere
	606	Not Acceptable

  SIP Trunk configuration on Cisco Voice Gateway

!

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  registrar server expires max 3600 min 3600
  outbound-proxy ipv4:xxx.xxx.xxx.xxx:5060

--- SIP UA Configuration ---

sip-ua
 keepalive target ipv4:xxx.xxx.xxx.xxx:5060
 authentication username xxxxxxx password 7  xxxxxx
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 timers keepalive active 10
 sip-server ipv4:4.xxx.xxx.xxx:5060
 host-registrar
!
voice class codec 1
 codec preference 1 g711ulaw
!
--- SIP Trunk Configuration ---

dial-peer voice 100 VoIP
 permission term
 description ** Incoming call from SIP trunk (Level 3) **
 session protocol sipv2
 session target sip-server
 incoming called-number .%
 voice-class codec 1 
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
no vad
!
Dial-peer voice 101 VoIP
Description ******outgoing call to SIP trunk******
Destination-pattern 9.T
 Session protocol sipv2
 Session target sip-server
 Voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
  dtmf-relay rtp-nte
no vad
!

Troubleshooting Tools

If calls cannot be made between SIP gateways or over SIP trunks, dial peer configuration is one of the first places to check. Make sure that the dial peer is configured to use SIP and that both devices are using the same transport protocol and DTMF relay method. Make sure that destination patterns and session targets are correct, also.

The show sip-ua connections {udp|tcp} command gives you information on active connections, including those with errors. To stop a problem connection, use the clear sip-ua {udp | tcp} [connection id number] [target ipv4: ip-address] command.

To ensure that the SIP is enabled on the gateway, use the show sip-ua service command. You should get the following result:

SIP-GW#show sip-ua service

SIP Service is up