Remote Audio not connecting: iOS, PJSIP 2.6, CallKit, PJSUA2 - ios

Question

I am updating an existing iOS VOIP application to use CallKit with PJSIP 2.6 and PJSUA2.
After some effort, the CallKit implementation seems to be working as expected. Incoming calls can be accepted or declined, and if accepted, will be connected and controlled with an in-app active call view controller.
The audio, however, does not appear to be properly connected at the pjsip end. There is no audio coming in from, or going out to the remote caller. The microphone audio appears to be routed back to the iPhone speaker.
The SIP audio ports should be connecting in callback function onCallMediaState:
virtual void onCallMediaState(OnCallMediaStateParam &prm) {
CallInfo ci = getInfo();
AudioMedia* audio_media = 0;
for (unsigned i = 0; i < ci.media.size(); i++) {
if (ci.media[i].type==PJMEDIA_TYPE_AUDIO && ( ci.media[i].status == PJSUA_CALL_MEDIA_ACTIVE ||
ci.media[i].status ==PJSUA_CALL_MEDIA_REMOTE_HOLD)) {
try {
audio_media = static_cast<AudioMedia*>(getMedia(i));
if(audio_media != 0)
{
Endpoint::instance().audDevManager().getCaptureDevMedia().startTransmit(*audio_media);
audio_media->startTransmit(Endpoint::instance().audDevManager().getPlaybackDevMedia());
}
} catch (std::exception ex) {
continue;
}
}
}
}
As described in Ticket#1941 at:
https://trac.pjsip.org/repos/ticket/1941:
I set the audio devices using:
ep->audDevManager().setNullDev();
immediately after the initialization of the Endpoint class (ep->libInit(epConfig);), and then:
I attempt to set the devices using pjsua_set_snd_dev() in CXProvider’s didActivate function, like this:
-(void) setSipSoundDevices {
pj_status_t status;
int captDev, playDev;
pjsua_get_snd_dev(&captDev, &playDev);
Endpoint::instance().audDevManager().setPlaybackDev(playDev);
Endpoint::instance().audDevManager().setCaptureDev(captDev);
}
pjsua_get_snd_dev(&captDev, &playDev) returns -99, -99 and the audio does not connect.
My question is this. How can I properly hook up the remote audio sources or ports, on an incoming call using PJSIP 2.6 and CallKit?
Might 2.5.5 work better in this regard?
Any insights are appreciated.

Answer 1

By and by I got the incoming call audio working properly. The crux of the matter was that even though the documentation from both Apple and SIP say that the audio has to be handled on the iOS end, you still have to set the SIP audio devices in the SIP layer in the provider delegate 'didActivate' and 'didDeactivate' functions. Because I use the PJSUA C++ layer, I had to drill down through the objc-c++ bridging layer to provide this functionality. ie.
-(void) activateSipSoundDevices {
pj_status_t status = pjsua_set_snd_dev(0, 0);
}
-(void) deactivateSipSoundDevices {
pj_status_t status = pjsua_set_null_snd_dev();
}
When initializing the SIP Account, be sure to set the null sound devices like:
ep->audDevManager().setNullDev();
Hope this helps.