SIP (Session Initiation Protocol) Call Flow

Hi All,

        We have already discussed the basics of SIP in our last post. Here we would like to share the SIP call flow. Here we have also included PSTNs, so that the reader can co-relate the message of SIP and ISUP.

Scenario:
   
     A Number wants to call  B Number which is catered by PSTN B.

• PSTN A to MSS X protocol used in ISUP.
• MSS X to MSS Y protocol used in SIP.
• MSS Y to PSTN B protocol is ISUP.

1) After receiving IAM  from PSTN A with called party number as B number  MSS X after number analysis detects the B number has to be routed to MSS Y which is connected by SIP. MSS X send a INVITE message.
Major components of INVITE are "Called Party", "Calling Party", "Bearer information", "Codec".

2) MSS Y receives INVITE and responds with 100 (Trying) message.
This response indicates that the request has been received by the next-hop server and that some unspecified action is being taken on behalf of this call (for example, a database is being consulted). At the same time after number analysis its send IAM to PSTN B.

3) MSS Y send 183 message to MSS X. This message is called as Session progress indicated that session is in progress. In response to 183 MSS A sends PRACK. PRACK (PRovisional ACKnowledgement) is like any another request within a dialog.
** PRACK is response for 1XX mesages
    ACK is response for 2XX messages.

4) After analysis of B number PSTN B sends ACM with "Called Party status indicator = no indication" to MSS Y. MSS Y sends 200 message to MSS X in turn MSS X forwards ACM message to PSTN A.

5) When B number starts ringing PSTN B send CPG message with "Called Party status indicator = Subscriber free". Indicates that subscriber is free and ringing.
MSS Y send 180 ringing message to MSS X.

Which is communicated to PSTN A in CPG message and A number can hear a ringtone.

6) B number answers the call in response PSTN B send ANM message to MSS Y. MSS Y send 200(With ANM) message to MSS X. MSS X responds with ACK , also forwards ANM message to PSTN A. At this point of time speech path is through.

7) After conversation A number releases the call. REL is send from PSTN A to MSS X in response MSS X sends BYE message to MSS Y. MSS Y send the REL message to PSTN B.

8) PSTN B release the resources and respond with RLC message to MSS Y. MSS Y relays the 200(with RLC) message to MSS X. RLC is then forwarded to PSTN A. This complete the release of all resources used for call.

Comments and Appreciations are most welcome.

ChEEeeeEEeeeErs!!!
Telecom Tigers Team
telecomtigers@gmail.com
http://homepageforu.webs.com/

SIP Messages & Release Causes

Hi All,
             As we all know, now a days all the services in telecom are moving into IP domain, in which "Session Initiation Protocol (SIP)" plays a vital role, so here we go for SIP Call flow, but before that, we need some basics for the same,

SIP :- It's a text based protocol responsible for the establishment, management and tearing down of media sessions in an Internet Protocol (IP) environment.

SIP Messages :- It was designed using a request/response model, there are 2 types of SIP messages – request (method) and responses.

Request :- “A SIP message sent from a client to a server, for the purpose of invoking a particular operation. There are different requests – Invite, Register, Bye, ACK, Cancel and Options (Refer, Subscribe, Notify, Publish, Message Update, Info and PRACK)

Responses :- “A SIP message sent from a server to a client, to indicate the status of a request sent from the client to the server.” Responses are differentiated into 6 classes.

Response Class	Phase
1XX (Informational)	100 (Trying)
	180 (Ringing)
	181 (Call is being Forwarded)
	182 (Call Queued)
	183 (Session Progress)
2XX (Success)	200 (OK)
	202 (Accepted)
	204 (No Notification)
3XX (Redirection)	300 (Multiple Choices)
	301 (Moved Permanently)
	302 (Moved Temporarily)
	305 (Use Proxy)
	380 (Alternative Service)
4XX (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)
	409 (Conflict)
	410 (Gone)
	411 (Length Required)
	412 (Conditional Request Failed)
	413 (Request Entity Too Large)
	414 (Request-URI Too Long)
	415 (Unsupported Media Type)
	416 (Unsupported URI Scheme)
	417 (Unknown Resource Priority)
	420 (Bad Extension)
	421 (Extension Required)
	422 (Session Interval Too Short)
	423 (Interval Too Brief )
	428 (Use Identity Header )
	429 (Provide Referrer Identity)
	430 (Flow Failed)
	433 (Anonymity Disallowed)
	436 (Bad Identity-Info Header )
	437 (Unsupported Certificate)
	438 (Invalid Identity Header )
	439 (First Hop Lacks Outbound Support)
	440 (Max-Breadth Exceeded)
	470 (Consent Needed)
	480 (Temporarily Unavailable)
	481 (Dialog/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)
	489 (Bad Event)
	491 (Request Pending)
	493 (Request Undecipherable)
	494 (Security Agreement Required)
5XX (Server Error)	500 (Server Internal Error )
	501 (Not Implemented)
	502 (Bad Gateway)
	503 (Service Unavailable)
	504 (Gateway Timeout)
	505 (Version Not Supported)
	513 (Message Too Large)
	580 (Preconditions Failure)
6XX (Global Error)	600 (Busy Everywhere)
	603 (Decline)
	604 (Does Not Exist Anywhere)
	606 (Not Acceptable)

One of the most significant advantages of SIP trunking is its ability to combine data, voice and video in a single line, eliminating the need for separate physical media for each mode.

SIP call flow will be shared in coming month.

Till then happy reading, comments and appreciations are most welcome.

ChEEEEErs!!!
Telecom Tigers Team
telecomtigers@gmail.com
http://homepageforu.webs.com/

What is SIP

SIP (Session Initiation Protocol) :-

It is one of the signaling protocol as SS7 & H.248,

It is an application layer protocol that can extablish, modify & terminate sessions or calls. These sessions include multimedia conference, internet telephony, & similiar applications.

SIP is one of the key protocol that implements voice-over IP (VOIP).

It is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.

SIP supported services -

• Name Mapping
• Redirection
• ISDN Services
• Intelligent Network (IN) services.
• User location
• User capabilities
• User availability
• Call set-up
• Call handling
• Call forwarding
• Call-forwarding no answer
• Call-forwarding busy
• Call-forwarding unconditional
• Other address-translation services
• Callee and calling "number" delivery, where numbers can be any (preferably unique) naming scheme
• Personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals
• Terminal-type negotiation and selection: a caller can be given a choice how to reach the party, e.g. via Internet telephony, mobile phone, an answering service, etc.
• Terminal capability negotiation
• Caller and callee authentication
• Blind and supervised call transfer
• Invitations to multicast conference

It support 5 facets of extablishing & terminating multimedia communications :-

1. User Location - Detemining end system to be used for communication.
2. User Capabilities - Determining the media & media parameters tobe used.
3. User Availability - Determining the willingness of called party to engange in communication.
4. Call Setup - Sending ringback tone to the called party & estabilishing call parameters at both end called & calling party.   
5. Call Holding & Control - Includes redirection, transfer & termination of calls.

SIP can also initiate multi-party calls using multipoint control unit (MCU) or fully meshed interconnection.
Internet Telephony gateway that connects PSTN parties can also use SIP to setup calls between them.
SIP can use User Datagram Protocol (UDP) & Transmision Control Protocol (TCP) as transport protocol, UDP is preferred.

Architecture
There are two basic components within SIP:

• SIP user agent.
• SIP network server.

The User Agent is the end system component for the call. The user agent itself has a client element, the User Agent Client (UAC) and a server element, the User Agent Server (UAS). The client element initiates the calls and the server element answers the calls. This allows peer-to-peer calls to be made using a client-server protocol.

SIP user agents can be lightweight clients suitable for embedding in end-user devices such as mobile handsets or PDAs. Alternatively, they can be desktop applications that bind with other software applications such as contact managers.


The SIP server is the network device that handles the signalling associated with multiple calls. The main function of the SIP servers is to provide name resolution and user location, since the caller is unlikely to know the IP address or host name of the called party, and to pass on messages to other servers using next hop routing protocols.

SIP servers can operate in two different modes: stateful and stateless. The difference between these modes is that a server in a stateful mode remembers the incoming requests it receives, along with the responses it sends back and the outgoing requests it sends on.

A server acting in a stateless mode forgets all information once it has sent a request. These stateless servers are likely to be the backbone of the SIP infrastructure while stateful-mode servers are likely to be the local devices close to the user agents, controlling domains of users.

 
SIP Addressing :-
Uniform Resource Locator (URL) are used within SIP messages to indicate the originator (FROM), current destination (requested URL), final destination (TO) of a SIP request & to specify redirection address (Contact).

SIP URL has a Syntax :-

SIP:User:password@host:port;transport-param|user-param|method-param|ttl-param|maddr-param|other-param

Their meaning -

• SIP - indicates SIP is used for communication with a specified end system.
• User - Consists of any characters in the form of email address or telephone number.
• Password - can be included but not recommended because of security risk.
• Host - can be host(other user) domain name or IP address.
• Port - indicates port number to which request is sent, default is 5060, a public SIP port number.
• Transport-Param - Indicates which transport protocol to be used, TCP or UDP, default is UDP.
• User-Param - can be a telephone number, 2 values are available for this field, IP & Phone number, when field is set to "phone" username is telephone number & corresponding end system is an IP Telephony Gateway. 
• Method-Param - Specifies method or operation to be used.
• TTL-Param - Designates the Time-To-Live (TTL) of UDP multicast data packet. It is valid only when transport parameter is UDP & Maddr parameter is "Multicast Address".
• Maddr-Param - Provides the server address to be contacted for a user, overriding the address supplied in the host field. This address is typically a multicast address.

NOTE - The following parameters are optional
Transport-Param, User-Param, Method-Param, TTL-Param, Maddr-Param, Other-Param.


Thanks
telecomtigers@gmail.com
http://homepageforu.webs.com/