PJSIP学习笔记——PJSUA层发起呼叫的主要流程

在上一篇学习笔记从simple_pjsua.c示例程序了解PJSUA-LIB的基本使用流程中，使用了PJSUA层的
pjsua_call_make_call来发起一个呼叫，那么这个发起呼叫的流程是怎样的呢？先来看看这个函数：

/*
 * Make outgoing call to the specified URI using the specified account.
 */
PJ_DEF(pj_status_t) pjsua_call_make_call(pjsua_acc_id acc_id,
					 const pj_str_t *dest_uri,
					 const pjsua_call_setting *opt,
					 void *user_data,
					 const pjsua_msg_data *msg_data,
					 pjsua_call_id *p_call_id)
{
    pj_pool_t *tmp_pool = NULL;
    pjsip_dialog *dlg = NULL;
    pjsua_acc *acc;
    pjsua_call *call;
    int call_id = -1;
    pj_str_t contact;
    pj_status_t status;

    /* Check that account is valid */
    PJ_ASSERT_RETURN(acc_id>=0 || acc_id<(int)PJ_ARRAY_SIZE(pjsua_var.acc), 
		     PJ_EINVAL);

    /* Check arguments */
    PJ_ASSERT_RETURN(dest_uri, PJ_EINVAL);

    PJ_LOG(4,(THIS_FILE, "Making call with acc #%d to %.*s", acc_id,
	      (int)dest_uri->slen, dest_uri->ptr));

    pj_log_push_indent();

    PJSUA_LOCK();

//    创建声音设备
    /* Create sound port if none is instantiated, to check if sound device
     * can be used. But only do this with the conference bridge, as with 
     * audio switchboard (i.e. APS-Direct), we can only open the sound 
     * device once the correct format has been known
     */
    if (!pjsua_var.is_mswitch && pjsua_var.snd_port==NULL && 
	pjsua_var.null_snd==NULL && !pjsua_var.no_snd) 
    {
	status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev);
	if (status != PJ_SUCCESS)
	    goto on_error;
    }

//    检查SIP帐号
    acc = &pjsua_var.acc[acc_id];
    if (!acc->valid) {
	pjsua_perror(THIS_FILE, "Unable to make call because account "
		     "is not valid", PJ_EINVALIDOP);
	status = PJ_EINVALIDOP;
	goto on_error;
    }

//    创建呼叫标识
    /* Find free call slot. */
    call_id = alloc_call_id();

    if (call_id == PJSUA_INVALID_ID) {
	pjsua_perror(THIS_FILE, "Error making call", PJ_ETOOMANY);
	status = PJ_ETOOMANY;
	goto on_error;
    }

//    复位呼叫参数
    /* Clear call descriptor */
    reset_call(call_id);

    call = &pjsua_var.calls[call_id];

    /* Associate session with account */
    call->acc_id = acc_id;
    call->call_hold_type = acc->cfg.call_hold_type;

//    设置呼叫参数
    /* Apply call setting */
    status = apply_call_setting(call, opt, NULL);
    if (status != PJ_SUCCESS) {
	pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
	goto on_error;
    }

    /* Create temporary pool */
    tmp_pool = pjsua_pool_create("tmpcall10", 512, 256);

    /* Verify that destination URI is valid before calling 
     * pjsua_acc_create_uac_contact, or otherwise there  
     * a misleading "Invalid Contact URI" error will be printed
     * when pjsua_acc_create_uac_contact() fails.
     */
    if (1) {
	pjsip_uri *uri;
	pj_str_t dup;

//        分析被叫SIP号码
	pj_strdup_with_null(tmp_pool, &dup, dest_uri);
	uri = pjsip_parse_uri(tmp_pool, dup.ptr, dup.slen, 0);

	if (uri == NULL) {
	    pjsua_perror(THIS_FILE, "Unable to make call", 
			 PJSIP_EINVALIDREQURI);
	    status = PJSIP_EINVALIDREQURI;
	    goto on_error;
	}
    }

    /* Mark call start time. */
    pj_gettimeofday(&call->start_time);

    /* Reset first response time */
    call->res_time.sec = 0;

//    创建Contact头域
    /* Create suitable Contact header unless a Contact header has been
     * set in the account.
     */
    if (acc->contact.slen) {
	contact = acc->contact;
    } else {
	status = pjsua_acc_create_uac_contact(tmp_pool, &contact,
					      acc_id, dest_uri);
	if (status != PJ_SUCCESS) {
	    pjsua_perror(THIS_FILE, "Unable to generate Contact header", 
			 status);
	    goto on_error;
	}
    }

//    创建SIP对话（Dialog）
    /* Create outgoing dialog: */
    status = pjsip_dlg_create_uac( pjsip_ua_instance(), 
				   &acc->cfg.id, &contact,
				   dest_uri, dest_uri, &dlg);
    if (status != PJ_SUCCESS) {
	pjsua_perror(THIS_FILE, "Dialog creation failed", status);
	goto on_error;
    }

    /* Increment the dialog's lock otherwise when invite session creation
     * fails the dialog will be destroyed prematurely.
     */
    pjsip_dlg_inc_lock(dlg);

//    设置Via头域
    if (acc->cfg.allow_via_rewrite && acc->via_addr.host.slen > 0)
        pjsip_dlg_set_via_sent_by(dlg, &acc->via_addr, acc->via_tp);

//    设置安全级别，安全级别有何作用？
    /* Calculate call's secure level */
    call->secure_level = get_secure_level(acc_id, dest_uri);

//    设置用户数据，用户数据是什么？
    /* Attach user data */
    call->user_data = user_data;
    
//    复制消息数据，消息数据有何作用？
    /* Store variables required for the callback after the async
     * media transport creation is completed.
     */
    if (msg_data) {
	call->async_call.call_var.out_call.msg_data = pjsua_msg_data_clone(
                                                          dlg->pool, msg_data);
    }
//    保存对话信息
    call->async_call.dlg = dlg;

    /* Temporarily increment dialog session. Without this, dialog will be
     * prematurely destroyed if dec_lock() is called on the dialog before
     * the invite session is created.
     */
    pjsip_dlg_inc_session(dlg, &pjsua_var.mod);

//    初始化媒体通道
    /* Init media channel */
    status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC, 
				      call->secure_level, dlg->pool,
				      NULL, NULL, PJ_TRUE,
                                      &on_make_call_med_tp_complete);
//    调用媒体传输回调函数
    if (status == PJ_SUCCESS) {
        status = on_make_call_med_tp_complete(call->index, NULL);
        if (status != PJ_SUCCESS)
	    goto on_error;
    } else if (status != PJ_EPENDING) {
	pjsua_perror(THIS_FILE, "Error initializing media channel", status);
        pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
	goto on_error;
    }

    /* Done. */

    if (p_call_id)
	*p_call_id = call_id;

    pjsip_dlg_dec_lock(dlg);
    pj_pool_release(tmp_pool);
    PJSUA_UNLOCK();

    pj_log_pop_indent();

    return PJ_SUCCESS;

on_error:
    if (dlg) {
	/* This may destroy the dialog */
	pjsip_dlg_dec_lock(dlg);
    }

    if (call_id != -1) {
	pjsua_media_channel_deinit(call_id);
	reset_call(call_id);
    }

    pjsua_check_snd_dev_idle();

    if (tmp_pool)
	pj_pool_release(tmp_pool);
    PJSUA_UNLOCK();

    pj_log_pop_indent();
    return status;
}

我们先来看看如何分配一个呼叫标识：

/* Allocate one call id */
static pjsua_call_id alloc_call_id(void)
{
    pjsua_call_id cid;

#if 1
    /* New algorithm: round-robin */
    if (pjsua_var.next_call_id >= (int)pjsua_var.ua_cfg.max_calls || 
	pjsua_var.next_call_id < 0)
    {
	pjsua_var.next_call_id = 0;
    }

//    从next_call_id到max_calls之间找一个空闲的calls数组元素
    for (cid=pjsua_var.next_call_id;
	 cid<(int)pjsua_var.ua_cfg.max_calls; 
	 ++cid) 
    {
	if (pjsua_var.calls[cid].inv == NULL &&
            pjsua_var.calls[cid].async_call.dlg == NULL)
        {
	    ++pjsua_var.next_call_id;
	    return cid;
	}
    }

//    从0到next_call_id之间找一个空闲的calls数组元素
    for (cid=0; cid < pjsua_var.next_call_id; ++cid) {
	if (pjsua_var.calls[cid].inv == NULL &&
            pjsua_var.calls[cid].async_call.dlg == NULL)
        {
	    ++pjsua_var.next_call_id;
	    return cid;
	}
    }

#else
    /* Old algorithm */
    for (cid=0; cid<(int)pjsua_var.ua_cfg.max_calls; ++cid) {
	if (pjsua_var.calls[cid].inv == NULL)
	    return cid;
    }
#endif

    return PJSUA_INVALID_ID;
}

从上面的函数来看，这里的分配呼叫标识只是在calls数据中寻找一个空闲的单元（用于存放呼叫数据），这个呼叫标识并不是SIP协议里面的CALL ID的概念。

reset_call函数就是将呼叫参数设置为0值：

*
 * Reset call descriptor.
 */
static void reset_call(pjsua_call_id id)
{
    pjsua_call *call = &pjsua_var.calls[id];
    unsigned i;

    pj_bzero(call, sizeof(*call));
    call->index = id;
    call->last_text.ptr = call->last_text_buf_;
    for (i=0; i<PJ_ARRAY_SIZE(call->media); ++i) {
	pjsua_call_media *call_med = &call->media[i];
	call_med->ssrc = pj_rand();
	call_med->strm.a.conf_slot = PJSUA_INVALID_ID;
	call_med->strm.v.cap_win_id = PJSUA_INVALID_ID;
	call_med->strm.v.rdr_win_id = PJSUA_INVALID_ID;
	call_med->call = call;
	call_med->idx = i;
	call_med->tp_auto_del = PJ_TRUE;
    }
    pjsua_call_setting_default(&call->opt);
    pj_timer_entry_init(&call->reinv_timer, PJ_FALSE,
			(void*)(pj_size_t)id, &reinv_timer_cb);
}

设置呼叫参数：

static pj_status_t apply_call_setting(pjsua_call *call,
				      const pjsua_call_setting *opt,
				      const pjmedia_sdp_session *rem_sdp)
{
    pj_assert(call);

    if (!opt)
	return PJ_SUCCESS;

#if !PJMEDIA_HAS_VIDEO
    pj_assert(opt->vid_cnt == 0);
#endif

    call->opt = *opt;

//    如果呼叫已建立，则设置本端的对话角色
//    如果有远端SDP，则本端为UAS（User Agent Server），否则为UAC
    /* If call is established, reinit media channel */
    if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) {
	pjsip_role_e role = rem_sdp? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC;
	pj_status_t status;

//        初始化媒体通道
	status = pjsua_media_channel_init(call->index, role,
					  call->secure_level,
					  call->inv->pool_prov,
					  rem_sdp, NULL,
					  PJ_FALSE, NULL);
	if (status != PJ_SUCCESS) {
	    pjsua_perror(THIS_FILE, "Error re-initializing media channel",
			 status);
	    return status;
	}
    }

    return PJ_SUCCESS;
}