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Asterisk Source/Sip Module


* call-limit

* static int load_module(void)
    • user, peer, registry object list
  • sched = sched_context_create()
  • io = io_context_create()
  • reload_config(sip_reloadreason)
  • ast_channel_register(&sip_tech)
  • register
    • cli, rtp_proto, udptl_prot, application, custom_function, manager
  • sip_poke_all_peers();
  • sip_send_all_registers();
  • restart_monitor();

* static int restart_monitor(void)
  • monitor_thread == pthread_self()
  • ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL)
  • return 0;

* do_monitor
  • sipsock_read
    • handle_request
      • handle_response

 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.


* static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
  • if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
    • Can't find codec to connect to host
    • return NULL;
  • }
  • if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
  • }
  • if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
    • return NULL;
  • }
  • if (create_addr(p, host, NULL)) {
    • return NULL;
  • }
  • tmpc = sip_new(p, AST_STATE_DOWN, host);
  • restart_monitor();
  • return tmpc;

* static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title)
  • {
    • tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i);
  • }
  • if (!tmp) {
    • return NULL;
  • }
  • if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
    • Unable to start PBX on
    • tmp = NULL;
  • }
  • return tmp;

* ast_pbx_start

* static int sip_call(struct ast_channel *ast, char *dest, int timeout)
  • if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
    • return -1;
  • }
  • res = update_call_counter(p, INC_CALL_RINGING);
  • if ( res != -1 ) {
    • if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
      • res = -1;
    • } else {
      • xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
      • if (xmitres == XMIT_ERROR)
        • return -1;
      • p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
    • }
  • } else {
    • ast->hangupcause = AST_CAUSE_USER_BUSY;
  • }
  • return res;

Sip protocol

* What are the benefits of SIP?
  • Generally it is used by two end-points to negotiate a "call."
  • By negotiate I mean
    • the medium (text, voice, other),
    • the transport (usually RTP, Real Time Protocol), and
    • the encoding (codec).
  • Once the negotiation is successful,
    • the two end-points use the selected method
    • for talking to each other - independently of SIP.
  • Once the "call" is over,
    • SIP is used to indicate a disconnection.
  • Therefore, SIP is best used as a signaling mechanism.
  • SIP and its extensions also provide related functions
    • such as instant messaging, registration, and presence.

* A sip call session between 2 phones is established as follows:
  • The calling phone sends out an invite
  • The called phone sends an information response 100 - Trying - back.
  • When the called phone starts ringing a response 180 - Ringing - is sent back
  • When the caller picks up the phone, the called phone sends a response 200 - OK
  • The calling phone responds with ACK - acknowledgement
  • Now the actual conversation is transmitted as data via RTP
  • When the person calling hangs up, a BYE request is sent to the calling phone
  • The calling phone responds with a 200 - OK.

* Call-ID
  • The Call-ID is seen as a unique identifier for a particular call.
  • It is then used by the originating client to match responses appropriately.
  • Call-ID acts as a unique identifier to group together a series of messages.
  • So what does this group means?
    • A Sip call consists of a series of requests/responses.
    • Clients (both client and server) need a mechanism to associate(group) such requests/responses.
* Components of sip
  • User Agent Client (UAC)
  • User Agent Server (UAS)
    • Proxy server
    • Redirect server
    • Registra
    • Location server
* SIP request and reply
  • start line
    • request INVITE sip:user@sipserver.com SIP/2.0
    • reply SIP/2.0 200 OK
  • message headers
    • Via
    • From
    • To
    • Call-id
    • Cseq
    • Contact
    • User-Agent
    • Content-Type
    • Content-Length
  • message body
* Request methods
  • INVITE : Indicates that the user or service is being invited to participate in a session. The body of this message would include a description of the session to which the callee is being invited.
  • ACK : Confirms that the client has received a final response to an INVITE request, and is only used with INVITE requests.
  • BYE : Is sent by a User Agent Client to indicate to the server that it wishes to terminate the call.
  • CANCEL : Is used to cancel a pending request.
  • OPTIONS : Is used to query a server about its capabilities.
  • REGISTER : Is used by a client to register an address with a SIP server.
  • INFO
* Response classes
  • provisional
  • final


* A CSeq header field in a request contains
  • a single decimal sequence number
  • and the request method
* The CSeq header field serves
  • to identify and order transactions within a dialog,
  • to provide a means to uniquely identify transactions,
  • and to differentiate between new requests and request retransmissions.
* Two CSeq header fields are considered equal
  • if the sequence number and the request method are identical.

* Method
  • The method part of CSeq is case-sensitive and MUST match that of the request.
* Sequence number
  • is chosen by the requesting client
  • and is unique within a single value of Call-ID.
  • MUST be expressible as a 32-bit unsigned integer
  • and MUST be less than 2**31.
  • For non-REGISTER requests outside of a dialog,
    • the sequence number value is arbitrary.
  • Consecutive Requests that differ in method, headers or body, but have the same CallIdHeader
    • must contain strictly monotonically increasing
    • and contiguous sequence numbers;
    • sequence numbers do not wrap around.
  • Retransmissions of the same Request carry the same sequence number,
  • but an INVITE Request with a different message body or different headers (a "re-invitation") acquires a new, higher sequence number.
  • A server must echo the CSeqHeader from the Request in its Response.
  • If the method value is missing in the received CSeqHeader,
    • the server fills it in appropriately.
  • ACK and CANCEL Requests must contain the same CSeqHeader sequence number (but not method) as the INVITE Request they refer to,
  • while a BYE Request cancelling an invitation must have a higher sequence number.
  • An user agent server must remember the highest sequence number for any INVITE Request with the same CallIdHeader.
  • The server must respond to, and then discard, any INVITE Request with a lower sequence number.

sip address

* sip address
  • A SIP address is a way to be reachable and to reach people. You can compare it to an e-mail address. You can signup for a free account on Ekiga.net. It will give you a unique SIP address that you can give to your friends so that they can contact you. An example of SIP address is sip:dsandras@ekiga.net.
* sip
  • SIP is the Session Initiation Protocol, defined in RFC 3261. It is the most widely used standard for VoIP signalling. It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants. SIP is used during and after the call setup.
* sip URI
  • URI Means "Uniform Resource Identifier". A SIP URI is used to identify an entity to be addressed via SIP, just like an email address identifies the email recipient. The general format of a SIP URI is user@domain:port user" identifies the particular user to address. The user can be a phone number, if this is supported by the SIP server.
Your ¡°SIP program¡± registers its online presence with a ¡°SIP Proxy¡±: ¡°Hey, i¡¯m on my home network right now, and I can be reached at this IP address¡±. When you arrive at work, your SIP program will now say ¡°Yoohoo, i¡¯ve moved, i¡¯m now here!¡±.

If someone wants to call you, they¡¯ll type your SIP address in their SIP program. The SIP provider will help this person¡¯s SIP software get in touch with your computer¡¯s SIP software, partly thanks to some STUN magic thrown in the middle. A SIP address looks exactly like an e-mail address, and, with some providers such as EarthLink, can very-well be one and the same. In my case, you can send me an e-mail at hollandct@earthlink.net or plug hollandct@earthlink.net ( or sip:hollandct@earthlink.net ) in your SIP program to call me up. If i¡¯m not online or available, you¡¯ll hear my voicemail, which will then be delivered as a .wav attachment to my e-mail address ¡¦ which Mail.app plays inline just fine!

You don¡¯t even need a ¡°SIP Provider¡± to do SIP. If you know your party¡¯s IP address or host name, if their SIP software is properly configured, you can plug their IP address into your SIP program to give them a ring.

Having a SIP address just gives you a more universal way for people to get in touch with you.

Standing on head makes smile of frown, but rest of face also upside down.

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