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

Question

I am using PJSIP 2.0,Xcode 7.3 and ios 9.2.1, Suppose we have 3 users A,B & C user.
When ever A call to B (its working fine ) but when A add member C in call (as conference call) then A & B can communicate with C but C never get Voice from B .(C & A communicate fine ).So how to manage communication between C & B .

Answer 1

This code working fine for me , only need to merge pjsua_conf_connect() slot number 0 1 and , which is our requirement.

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

You can use pjsip ios library 2.5 latest build and another call. Second is using webservice to originate call and redirect channels to confbridge for asterisk as sip server.

If you are using another media server then share the details as well.