和以前版本相比，Asterisk在架构上有了不小的变动，本文基于asterisk 1.8.10.1分析整理。

chan_sip模块属于通道驱动模块。它实现了协议的相关内容，使Asterisk能够和支持SIP协议的其它设备通信。在1.8版本下，还是没有实现S/MIME的内容，有部分代码实现了TCP和TLS，但我没用过。对SIP事务的支持，还是不好。

在chan_sip.c文件的顶部，简要地描述了这个模块实现的功能和一些缺陷，并描述了这个模块的发展计划，这里不重复这些内容。

说明，下面代码引用处的行号来源于asterisk社区doxygen生成的文档，因为代码更新同步原因，可能和您看到的实际代码略有差异

模块初始化

Asterisk是模块化设计的，内核会负责管理外围的模块。内核管理模块信息的回调函数，封装在ast_module_info这个数据结构中。这个数据结构的实例化过程，定义了一个宏，叫做AST_MODULE_INFO，所有的外围模块都会调用这个宏。在chan_sip.c中，实例化的代码是

31889 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",

31890       .load = load_module,

31891       .unload = unload_module,

31892       .reload = reload,

31893       .load_pri = AST_MODPRI_CHANNEL_DRIVER,

31894       .nonoptreq = "res_crypto,chan_local",

31895           );

上面说明了模块的加载函数入口load_module。当内核加载SIP模块时，就会跳转到这里执行。这个函数的原型是int load_module(void)。

load_module首先初始化一系列的容器，1.8版本里，容器采用哈希表，不过这个表不能自动调整大小。容器实例主要有peers、dialogs等，这些在模块范围内是全局的。

接下来，创建调度管事器和POLL模型的IO管理器。接下来，调用reload_config(sip_reloadreason)读取模块的配置文件。

31600    can_parse_xml = sip_is_xml_parsable();

31601    if (reload_config(sip_reloadreason)) { /* Load the configuration from sip.conf */

31602       return AST_MODULE_LOAD_DECLINE;

31603    }

UDP是缺省支持的，如果配置文件里配置了TCP或TLS支持，那么在reload_config里会初始化相应的IO。

接下来，注册通道类型和一系列的回调函数入口，比如说调用ast_rtp_glue_register设置RTP引擎相关的回调函数。最后，调用restart_monitor()来创建SIP监听线程。restart_monitor还在另外两个地方被调用，一个是sip_request_call，一个是sip_reload。

在restart_monitor函数中，调用ast_pthread_create_background创建一个新的线程，线程ID记录在monitor_thread，线程执行体是do_monitor。

26939       /* Start a new monitor */

26940       if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {

26941          ast_mutex_unlock(&monlock);

26942          ast_log(LOG_ERROR, "Unable to start monitor thread.\n");

26943          return -1;

26944       }

do_monitor首先调用ast_io_add添加一个IO实体，把IO的FD记录在sipsock_read_id，并注册IO回调函数sipsock_read，这个回调是SIP信令的入口。然后进入线程循环体。循环中，首先检查是否需要重新加载模块，然后检查容器中需要挂断的通话(比如说RTP监测超时)。

现在，回头来看reload_config这个函数，这里关注一下TCP和TLS初始化，如果对这部分不关心，可以跳过：

TCP

30000    /* Start TCP server */

30001    if (sip_cfg.tcp_enabled) {

30002       if (ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {

30003          ast_sockaddr_copy(&sip_tcp_desc.local_address, &bindaddr);

30004       }

30005       if (!ast_sockaddr_port(&sip_tcp_desc.local_address)) {

30006          ast_sockaddr_set_port(&sip_tcp_desc.local_address, STANDARD_SIP_PORT);

30007       }

30008    } else {

30009       ast_sockaddr_setnull(&sip_tcp_desc.local_address);

30010    }

30011    ast_tcptls_server_start(&sip_tcp_desc);

30012    if (sip_cfg.tcp_enabled && sip_tcp_desc.accept_fd == -1) {

30013       /* TCP server start failed. Tell the admin */

30014       ast_log(LOG_ERROR, "SIP TCP Server start failed. Not listening on TCP socket.\n");

30015    } else {

30016       ast_debug(2, "SIP TCP server started\n");

30017    }

TLS

30019    /* Start TLS server if needed */

30020    memcpy(sip_tls_desc.tls_cfg, &default_tls_cfg, sizeof(default_tls_cfg));

30022    if (ast_ssl_setup(sip_tls_desc.tls_cfg)) {

30023       if (ast_sockaddr_isnull(&sip_tls_desc.local_address)) {

30024          ast_sockaddr_copy(&sip_tls_desc.local_address, &bindaddr);

30025          ast_sockaddr_set_port(&sip_tls_desc.local_address,

30026                      STANDARD_TLS_PORT);

30027       }

30028       if (!ast_sockaddr_port(&sip_tls_desc.local_address)) {

30029          ast_sockaddr_set_port(&sip_tls_desc.local_address,

30030                      STANDARD_TLS_PORT);

30031       }

30032       ast_tcptls_server_start(&sip_tls_desc);

30033       if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {

30034          ast_log(LOG_ERROR, "TLS Server start failed. Not listening on TLS socket.\n");

30035          sip_tls_desc.tls_cfg = NULL;

30036       }

30037    } else if (sip_tls_desc.tls_cfg->enabled) {

30038       sip_tls_desc.tls_cfg = NULL;

30039       ast_log(LOG_WARNING, "SIP TLS server did not load because of errors.\n");

30040    }

这两个处理，都用到了一个关键的结构体ast_tcptls_session_args，实例名字分别是sip_tcp_desc和sip_tls_desc，看一下它们是怎样实例化的。

02213 static struct ast_tcptls_session_args sip_tcp_desc = {

02214    .accept_fd = -1,

02215    .master = AST_PTHREADT_NULL,

02216    .tls_cfg = NULL,

02217    .poll_timeout = -1,

02218    .name = "SIP TCP server",

02219    .accept_fn = ast_tcptls_server_root,

02220    .worker_fn = sip_tcp_worker_fn,

02221 };

02223 /*! \brief The TCP/TLS server definition */

02224 static struct ast_tcptls_session_args sip_tls_desc = {

02225    .accept_fd = -1,

02226    .master = AST_PTHREADT_NULL,

02227    .tls_cfg = &sip_tls_cfg,

02228    .poll_timeout = -1,

02229    .name = "SIP TLS server",

02230    .accept_fn = ast_tcptls_server_root,

02231    .worker_fn = sip_tcp_worker_fn,

02232 };

跟踪一下工作回调函数sip_tcp_worker_fn，会发现，最终调用了handle_request_do函数处理读取的数据，无论传输层用什么协议，数据都是调用这个函数处理的。

内核接口

01612 /*! \brief Definition of this channel for PBX channel registration */

01613 struct ast_channel_tech sip_tech = {

01614    .type = "SIP",

01615    .description = "Session Initiation Protocol (SIP)",

01616    .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,

01617    .requester = sip_request_call,         /* called with chan unlocked */

01618    .devicestate = sip_devicestate,        /* called with chan unlocked (not chan-specific) */

01619    .call = sip_call,       /* called with chan locked */

01620    .send_html = sip_sendhtml,

01621    .hangup = sip_hangup,         /* called with chan locked */

01622    .answer = sip_answer,         /* called with chan locked */

01623    .read = sip_read,       /* called with chan locked */

01624    .write = sip_write,        /* called with chan locked */

01625    .write_video = sip_write,     /* called with chan locked */

01626    .write_text = sip_write,

01627    .indicate = sip_indicate,     /* called with chan locked */

01628    .transfer = sip_transfer,     /* called with chan locked */

01629    .fixup = sip_fixup,        /* called with chan locked */

01630    .send_digit_begin = sip_senddigit_begin,  /* called with chan unlocked */

01631    .send_digit_end = sip_senddigit_end,

01632    .bridge = ast_rtp_instance_bridge,        /* XXX chan unlocked ? */

01633    .early_bridge = ast_rtp_instance_early_bridge,

01634    .send_text = sip_sendtext,    /* called with chan locked */

01635    .func_channel_read = sip_acf_channel_read,

01636    .setoption = sip_setoption,

01637    .queryoption = sip_queryoption,

01638    .get_pvt_uniqueid = sip_get_callid,

01639 };

这些函数，实现了内核的回调接口，在模块初始化时，会把这些函数的入口注册给内核的相关管理器。

入呼处理流程

前面说过，sipsock_read是UDP SIP消息的入口。调用ast_recvfrom(recvfrom函数的封装)，从socket上读取数据包。把数据存储在一个sip_request结构体对象的data字段中，然后调用handle_request_do函数处理读取的数据，这里面，很多地方的request泛指了SIP消息，包括response，光看字面，很容易误解。

首先，解析SIP消息：

26367    if (parse_request(req) == -1) { /* Bad packet, can't parse */

26368       ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */

26369       return 1;

26370    }

SIP消息解析完毕之后，匹配SIP的方法，在asterisk里，把response消息也当成一种方法来处理。接着调用find_call检索消息对应的sip_pvt结构，如果原先没有，find_call函数中会创建一个新的，最终把指针返回。紧接着，通过pvt结构中的owner字段，判断消息是否经过权鉴，最后调用handle_incoming函数，对消息进一步的处理：

26371    req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlPart1));

26373    if (req->debug)

26374       ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");

26376    if (req->headers < 2) { /* Must have at least two headers */

26377       ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */

26378       return 1;

26379    }

26380    ast_mutex_lock(&netlock);

26382    /* Find the active SIP dialog or create a new one */

26383    p = find_call(req, addr, req->method); /* returns p with a reference only. _NOT_ locked*/

26384    if (p == NULL) {

26385       ast_debug(1, "Invalid SIP message - rejected , no callid, len %zu\n", ast_str_strlen(req->data));

26386       ast_mutex_unlock(&netlock);

26387       return 1;

26388    }

26390    /* Lock both the pvt and the owner if owner is present.  This will

26391     * not fail. */

26392    owner_chan_ref = sip_pvt_lock_full(p);

26394    copy_socket_data(&p->socket, &req->socket);

26395    ast_sockaddr_copy(&p->recv, addr);

26397    /* if we have an owner, then this request has been authenticated */

26398    if (p->owner) {

26399       req->authenticated = 1;

26400    }

26402    if (p->do_history) /* This is a request or response, note what it was for */

26403       append_history(p, "Rx", "%s / %s / %s", req->data->str, sip_get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlPart2));

26405    if (handle_incoming(p, req, addr, &recount, &nounlock) == -1) {

26406       /* Request failed */

26407       ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");

26408    }

下面看一下handle_incoming函数，它首先对消息的合法性做一些检查，如果是应答消息则走入应答消息处理分枝，否则，继续向下处理：

26064    if (req->method == SIP_RESPONSE)

......

26099          handle_response(p, respid, e + len, req, seqno);

这其中，还检查消息是否为重发消息，如果是，填充重发标识：

else if (p->icseq &&

26143          p->icseq == seqno &&

26144          req->method != SIP_ACK &&

26145          (p->method != SIP_CANCEL || p->alreadygone)) {

26146       /* ignore means "don't do anything with it" but still have to

26147          respond appropriately.  We do this if we receive a repeat of

26148          the last sequence number  */

26149       req->ignore = 1;

26150       ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %u, ours %u)\n", sip_methods[p->method].text, p->icseq, seqno);

26151    }

根据3261规范定义，检查消息的一些通用合法性之后，根据请求的method，调用各自的处理函数：

26199    /* Handle various incoming SIP methods in requests */

26200    switch (p->method) {

26201    case SIP_OPTIONS:

26202       res = handle_request_options(p, req, addr, e);

26203       break;

26204    case SIP_INVITE:

26205       res = handle_request_invite(p, req, debug, seqno, addr, recount, e, nounlock);

               ......

下面，把注意力集中在handle_request_invite函数。

首先，查看消息里的Supported和Require头域，看asterisk能否支持对方所要求的扩展。

23101    if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->invitestate != INV_TERMINATED && p->invitestate != INV_CONFIRMED) && ast_channel_state(p->owner) != AST_STATE_UP)

上面这个判断的处理就是检查SIP扩展的。

23150    if (!req->ignore && p->pendinginvite)

这个判断请求的处理情况，是否正在处理。

接下来，检查是否有Replaces头域，如果有，做呼叫转移处理，里面调用到一个叫handle_invite_replaces的函数：

23192    p_replaces = sip_get_header(req, "Replaces");

23193    if (!ast_strlen_zero(p_replaces)) {

23194       /* We have a replaces header */

             ........

接下来的处理，有两种情况，一个是re-invite，一个是初始的invite。这两种情况的处理，有相同的地方，又有差异。代码中先处理不同的部分，再处理相同的部分；先处理re-invite，再处理原始INVITE。

23339    if (!req->ignore) {

23340       int newcall = (p->initreq.headers ? TRUE : FALSE);

23342       if (sip_cancel_destroy(p))

23343          ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");

23344       /* This also counts as a pending invite */

23345       p->pendinginvite = seqno;

23346       check_via(p, req);

23348       copy_request(&p->initreq, req);     /* Save this INVITE as the transaction basis */

23349       if (sipdebug)

23350          ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);

23351       if (!p->owner) {  /* Not a re-invite */

23352          if (debug)

23353             ast_verbose("Using INVITE request as basis request - %s\n", p->callid);

23354          if (newcall)

23355             append_history(p, "Invite", "New call: %s", p->callid);

23356          parse_ok_contact(p, req);

23357       } else { /* Re-invite on existing call */

              ......

}}

这一段处理re-invite的情况，主要是关于SDP变化处理什么的。

23417    if (!p->lastinvite && !req->ignore && !p->owner) {

23418       /* This is a new invite */

23419       /* Handle authentication if this is our first invite */

23420       int cc_recall_core_id = -1;

23421       set_pvt_allowed_methods(p, req);

23422       res = check_user_full(p, req, SIP_INVITE, e, XMIT_RELIABLE, addr, &authpeer);

23423       if (res == AUTH_CHALLENGE_SENT) {

23424          p->invitestate = INV_COMPLETED;     /* Needs to restart in another INVITE transaction */

23425          goto request_invite_cleanup;

23426       }

               ......

这一段处理原始INVITE的情况。check_user_full这个函数调用，检查被叫是否是合法的用户，还完成了RPT引擎的初始化。调用栈是：check_user_full-->check_peer_ok-->dialog_initialize_rtp-->ast_rtp_instance_new(还可能是check_user_full直接调用dialog_initialize_rtp)。RTP引擎的处理是Asterisk构架中比较重大的变化，缺省使用asterisk的RTP栈(称之为引擎)，但允许用户嵌入自己的RTP栈。

check_user_full返回之后，权鉴通过，设置标识位，然后处理SDP，如果请求中不带SDP，则可能是3PCC的流程：

23450       /* We have a successful authentication, process the SDP portion if there is one */

23451       if (find_sdp(req)) {

23452          if (process_sdp(p, req, SDP_T38_INITIATE)) {

23453             /* Asterisk does not yet support any Content-Encoding methods.  Always

23454              * attempt to process the sdp, but return a 415 if a Content-Encoding header

23455              * was present after processing fails. */

23456             if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {

23457                transmit_response_reliable(p, "415 Unsupported Media type", req);

23458             } else {

23459                /* Unacceptable codecs */

23460                transmit_response_reliable(p, "488 Not acceptable here", req);

23461             }

23462             p->invitestate = INV_COMPLETED;

23463             sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

23464             ast_debug(1, "No compatible codecs for this SIP call.\n");

23465             res = INV_REQ_ERROR;

23466             goto request_invite_cleanup;

23467          }

23468       } else { /* No SDP in invite, call control session */

23469          ast_format_cap_copy(p->jointcaps, p->caps);

23470          ast_debug(2, "No SDP in Invite, third party call control\n");

23471       }

对于原始INVITE来说，很关键的跳转点就是创建新的channel处理：

23554          /* First invitation - create the channel.  Allocation

23555           * failures are handled below. */

23556          c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL), NULL);

sip_new这个函数用于创建一个SIP CHANNEL，这里是入呼的调用，外呼时则是在sip_request_call函数中调用，sip_request_call是内核接口的SIP实现。sip_new返回之后，调用build_route记录SIP消息的路由信息。

回到handle_request_invite函数，re-invite和原始invite的差异处理完之后，处理SST(SIPSession Timer)扩展的支持：

23594    /* Session-Timers */

23595    if ((p->sipoptions & SIP_OPT_TIMER) && !ast_strlen_zero(sip_get_header(req, "Session-Expires")))

......

处理完SST之后，判断自身是还是Attendedtransfer 或call pickup的目标，最后，处理CHANNEL状态，根据状态，选择合适的应答码，交发出SIP应答。

23784       switch(c_state) {

23785       case AST_STATE_DOWN:

23786          ast_debug(2, "%s: New call is still down.... Trying... \n", ast_channel_name(c));

23787          transmit_provisional_response(p, "100 Trying", req, 0);

23788          p->invitestate = INV_PROCEEDING;

23789          ast_setstate(c, AST_STATE_RING);

23790          if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */

23791             enum ast_pbx_result result;

23793             result = ast_pbx_start(c);

23795             switch(result) {

23796             case AST_PBX_FAILED:

23797                ast_log(LOG_WARNING, "Failed to start PBX :(\n");

23798                p->invitestate = INV_COMPLETED;

23799                transmit_response_reliable(p, "503 Unavailable", req);

23800                break;

23801             case AST_PBX_CALL_LIMIT:

23802                ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");

23803                p->invitestate = INV_COMPLETED;

23804                transmit_response_reliable(p, "480 Temporarily Unavailable", req);

23805                res = AUTH_SESSION_LIMIT;

23806                break;

23807             case AST_PBX_SUCCESS:

23808                /* nothing to do */

23809                break;

23810             }

                ......

如果是新的呼叫，则调用ast_pbx_start，在这里面启动一个独立的线程，接管这个CHANNEL，新创建的线程入口是pbx_thread函数，在这个函数里调用__ast_pbx_run函数，跳转执行拨号计划。

外呼流程

从SIP角度理解，Asterisk就是一个B2BUA，一路通话，在Asterisk内部需要两个UA，Asterisk负责把两个UA桥接在一块。一个UA对应了一个通道，上面分析了入呼(incoming)通道，接下来我们分析一下外呼(outgoing)通道的情况。

还是从ast_pbx_start说起：

05518    if (increase_call_count(c))

05519       return AST_PBX_CALL_LIMIT;

05521    /* Start a new thread, and get something handling this channel. */

05522    if (ast_pthread_create_detached(&t, NULL, pbx_thread, c)) {

05523       ast_log(LOG_WARNING, "Failed to create new channel thread\n");

05524       decrease_call_count();

05525       return AST_PBX_FAILED;

05526    }

这几行代码检查是否还有空闲处理能力(取决于配置)，如果有调用ast_pthread_create_detached启动一个线程，线程入口函数是pbx_thread。当然，这个线程是处理入呼通道的。

跳转到pbx_thread看一下都做了些什么？

05479 static void *pbx_thread(void *data)

05480 {

05481    /* Oh joyeous kernel, we're a new thread, with nothing to do but

05482       answer this channel and get it going.

05483    */

05484    /* NOTE:

05485       The launcher of this function _MUST_ increment 'countcalls'

05486       before invoking the function; it will be decremented when the

05487       PBX has finished running on the channel

05488     */

05489    struct ast_channel *c = data;

05491    /* Associate new PBX thread with a call-id */

05492    struct ast_callid *callid = ast_create_callid();

05493    ast_callid_threadassoc_add(callid);

05494    callid = ast_callid_unref(callid);

05496    __ast_pbx_run(c, NULL);

05497    decrease_call_count();

05499    pthread_exit(NULL);

05501    return NULL;

05502 }

这里调用了一个很关键的函数：__ast_pbx_run。

       /* Start by trying whatever the channel is set to */

       if (!ast_exists_extension(c, c->context, c->exten, c->priority,

               S_COR(c->caller.id.number.valid, c->caller.id.number.str, NULL))) {

               /* If not successful fall back to 's' */

               ast_verb(2, "Starting %s at %s,%s,%d failed so falling back to exten 's'\n", c->name, c->context, c->exten, c->priority);

               /* XXX the original code used the existing priority in the call to

                * ast_exists_extension(), and reset it to 1 afterwards.

                * I believe the correct thing is to set it to 1 immediately.

*/

               set_ext_pri(c, "s", 1);

               if (!ast_exists_extension(c, c->context, c->exten, c->priority,

                       S_COR(c->caller.id.number.valid, c->caller.id.number.str, NULL))) {

                       /* JK02: And finally back to default if everything else failed */

                       ast_verb(2, "Starting %s at %s,%s,%d still failed so falling back to context 'default'\n", c->name, c->context, c->exten, c->priority);

                       ast_copy_string(c->context, "default", sizeof(c->context));

在这里，首先调用ast_exists_extension查找拨号计划的入口，如果找到了，进入一个for循环，逐条执行拨号计划：

05141    for (;;) {

05142       char dst_exten[256]; /* buffer to accumulate digits */

05143       int pos = 0;      /* XXX should check bounds */

05144       int digit = 0;

05145       int invalid = 0;

05146       int timeout = 0;

05148       /* loop on priorities in this context/exten */

05149       while (!(res = ast_spawn_extension(c, ast_channel_context(c), ast_channel_exten(c), ast_channel_priority(c),

05150          S_COR(ast_channel_caller(c)->id.number.valid, ast_channel_caller(c)->id.number.str, NULL),

05151          &found, 1))) {

05152          if (!ast_check_hangup(c)) {

05153             ast_channel_priority_set(c, ast_channel_priority(c) + 1);

05154             continue;

5155              }

         ........

ast_spawn_extension这个函数调用pbx_extension_helper完成具体的动作：

05051 int ast_spawn_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid, int *found, int combined_find_spawn)

05052 {

05053    return pbx_extension_helper(c, NULL, context, exten, priority, NULL, callerid, E_SPAWN, found, combined_find_spawn);

05054 }

pbx_extension_helper调用pbx_find_extension查找系统中的extension：

04342    e = pbx_find_extension(c, con, &q, context, exten, priority, label, callerid, action);

找到之后，调用pbx_exec执行

04390          return pbx_exec(c, app, passdata);  /* 0 on success, -1 on failure */

到目前为止，处理的都还是入呼的流程，如果需要外呼，就是在拨号计划中执行到dial()应用，从这个应用跳转到外呼处理。

在app_dial模块加载时，注册了一个应用的回调入口：

03160    res = ast_register_application_xml(app, dial_exec);

而dial_exec则调用了dial_exec_full：

03029 static int dial_exec(struct ast_channel *chan, const char *data)

03030 {

03031    struct ast_flags64 peerflags;

03033    memset(&peerflags, 0, sizeof(peerflags));

03035    return dial_exec_full(chan, data, &peerflags, NULL);

03036 }

在dial_exec_full函数里：首先调用ast_request分配通道资源，最终调用的就是通道模块注册的回调函数，SIP模块就是sip_request_call函数。然后调用ast_call发起呼叫，对应最终调用SIP通道注册的回调函数sip_call。最后，调用wait_for_answer等待被叫方应答。被叫应答后调用ast_bridge_call桥接两个通道。

Asterisk 1.8 chan_sip模块代码分析

模块初始化

内核接口

入呼处理流程

外呼流程

猜你喜欢