Question

我正在使用PJSIP 2.0，Xcode 7.3和ios 9.2.1，假设我们有3个用户A，B＆amp; C用户。
什么时候A呼叫B（工作正常）但是当A在呼叫中添加成员C（作为电话会议）时，A＆amp; B可以与C通信，但C从不从B获得语音。（C＆amp; A沟通正常）。那么如何管理C＆amp; C之间的通信。 B。

Answer 1

这段代码对我来说很好，只需要合并pjsua_conf_connect（）插槽号0 1，这是我们的要求。

static void on_call_media_state(pjsua_call_id call_id)
    {
        pjsua_call_info ci;
      SiphonApplication *app = (SiphonApplication *)[SiphonApplication sharedApplication];

        pjsua_call_get_info(call_id, &ci);
    //    PJ_LOG(3,(THIS_FILE,"on_call_media_state status %d count %d",
    //      ci.media_status
    //      pjmedia_conf_get_connect_count()));

      /* FIXME: Stop ringback */
      sip_ring_stop([app pjsipConfig]); 

      /* Connect ports appropriately when media status is ACTIVE or REMOTE HOLD,
       * otherwise we should NOT connect the ports.
       */

        pjsua_call_media_status slotOne = ci.media_status;
      if (slotOne == PJSUA_CALL_MEDIA_ACTIVE ||
          slotOne == PJSUA_CALL_MEDIA_REMOTE_HOLD)
      {
        // When media is active, connect call to sound device.
        pjsua_conf_connect(ci.conf_slot, 0);
        pjsua_conf_connect(0, ci.conf_slot);

        pjsua_conf_adjust_rx_level(0, 3.0);
        pjsua_conf_adjust_tx_level(0, 5.0);


      }


          [[NSUserDefaults standardUserDefaults] setObject: @"CallIsRunning"forKey:@"CallIsRunning"];
        if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { //    When media is active, connect call to sound device.
            pjsua_conf_port_id slotOne = ci.conf_slot;
            //        pjsua_conf_connect(slotOne, 0);
            //        pjsua_conf_connect(0, slotOne);
            //mergeCalls=true;

            mergeCalls=false;

            int max=pjsua_call_get_count();
            if (max==2) {

                [[NSUserDefaults standardUserDefaults] setInteger:pjsua_call_get_count() forKey:@"callCountForCalling"];
                mergeCalls=true;
            }


          NSString *grpID=  [[NSUserDefaults standardUserDefaults] objectForKey:@"callerTypegroup"];
            // if ([[[NSUserDefaults standardUserDefaults] objectForKey:@"callerTypegroup"] length]>1)

            if (mergeCalls == true && grpID.length==0) {

                pjsua_conf_port_id slotTwo = pjsua_call_get_conf_port(activeCallID);
                pjsua_conf_connect(slotOne, slotTwo);
                pjsua_conf_connect(slotTwo, slotOne);

                // since the "activeCallID" is already  talking, its conf_port is already connected to "0" (and vice versa) ...

            } else {
                activeCallID = call_id;
            }
        } else if (ci.media_status == PJSUA_CALL_MEDIA_LOCAL_HOLD) {
            // … callSuspended(callID);
        }

    }

Answer 2

您可以使用pjsip ios library 2.5最新版本和另一个调用。其次是使用webservice来发起呼叫和重定向信道，以便将星号作为sip服务器进行confbridge。

如果您正在使用其他媒体服务器，请分享详细信息。

如何使用Asterisk iOS进行电话会议PJSIP（在电话会议中添加新成员）？

2 个答案: