Greetings and a question about PJSUA2

[pnogueira@xxxxxxxxxxxxxxx (Paulo Candolo Nogueira)]

Hello Guys, how are you doing?
This is my first e-mail to this group and I have a question about PJSUA2.

I created a test app using PJSUA2 for iOS using the most recent svn release (http://svn.pjsip.org/repos/pjproject/trunk - 4722M) , and I was able to go as far as completing a call. The problem is that I got no audio (IN or OUT), and sometimes, a lot of RCTP address changes (about 1/second). I tested on multiple test call sips.
Its probably not the old problem with audio on the simulator since I have another app made using ipjsua in which I get normal audio. (But that was made using 4688M release)

I'm relativelly new to SIP and PJSIP, and I was hoping you guys could help me figuring out what might be wrong. And at least guiding me on the right direction.
My code is very simple, and pretty much like the samples found on the repo:


    Endpoint ep;
    std::auto_ptr<MyAccount> account(new MyAccount);

    ep.libCreate();
    
    //Endpoing configuration
    EpConfig ep_cfg;
    ep_cfg.logConfig.level = 4;
    ep_cfg.uaConfig.userAgent = "pjsua2-hello";
    
    ep.libInit( ep_cfg );

    //Transport
    TransportConfig tcfg;
    tcfg.port = 5062;
    try {
        ep.transportCreate(PJSIP_TRANSPORT_UDP, tcfg);
    } catch (Error &err) {
        std::cout << err.info() << std::endl;
    }

    ep.libStart();

    //registration
    AccountConfig acfg;
    acfg.idUri = "sip:my at sip.net";
    acfg.regConfig.registrarUri = "sip:sip.net";
    AuthCredInfo cred("digest", "*", "account", 0, "password");
    acfg.sipConfig.authCreds.push_back( cred );
    account->create(a cfg);

    pj_thread_sleep(2000);

    Call *call = new MyCall(*account);
    CallOpParam prm(true);
    prm.opt.audioCount = 1;
    prm.opt.videoCount = 0;
    call->makeCall("sip:test at echosip.com", prm);




I've got a feeling I'm missing something very simple here, but I can not figure out what. 
Thank you very much.