Hi, Thanks for the detailed report. I'm guessing that you have problem with NAT, hence STUN should be enabled. The instruction to enable NAT in [1] was outdated (sorry for that, but I just fixed it), the option should be "--stun-srv" instead of "--use-stun1". For example, add this in your cmdline: --stun-srv stun.pjsip.org Best regards, ?Benny [1] https://trac.pjsip.org/repos/wiki/Audio_Problems/Getting_Around_Nat On Thu, Sep 30, 2010 at 4:27 AM, Pablo Nu?ez <nunezlopez at gmail.com> wrote: > Hi everybody, I'm?stuck trying to make pjsip 1.8 work with my project, so I > finally decide to ask for help, I usually read the mails I receive from the > pjsip mailing list and I can see that a lof of you guys have very good > experience, so I some of you can point me in the right direction. > My problem is this: > I have mi VOIP app for iPhone using pjsip 1.0 working really nice, so I > decide to migrate to pjsip 1.8 to add multitasking support and receiving > calls while de iPhone is locked. > I downloaded the code and compile it using ./configure-iphone added the > libraries to my project and run it, everything fine but Im unable to hear > voice in local speaker, the guy in the other side can hear me pretty well so > I decide to follow the Troubleshooting sound problems section in the pjsip > site using the pjsua app. > I run it, add new account with +a, registered successfully with SIP server > and made a call with the same problem. > These are my results for the checklist suggestions: > > Check that the correct device is being used. > Here i see: > pjsua_media.c? Opening sound device PCM at 8000/1/20ms > > while making the call and > > pjsua_media.c > > Closing sound device after idle for 1 seconds > > pjsua_media.c? Closing iPhone IO device sound playback device and iPhone IO > device sound capture device > > while hanging up the call. > > Check that no other application is using the devices. These isn't the case, > no other audio is been played. > > Check that speaker is functioning properly by Looping-back Microphone to > Speaker device. The speaker is working properly because if change de include > folder path to the one holding the pjsip 1.0 version everything works fine. > > You can also check by Playing a WAV File to Speaker. I haven't tried this > but I think it should work for the same reason that checklist number 3, but > I always set the Volume level to max. > Check that Incoming RTP Packets are Indeed Received by Local Host. Indeed no > RTP packets are been received this is what I saw in the log while using dq > command: > > [CONFIRMED] To: sip:349201031 at abc.domain.com;tag=lignup-3896-4ca3a0a3 > Call time: 00h:01m:09s, 1st res in 4715 ms, conn in 6238ms > SRTP status: Not active Crypto-suite: (null) > #0 PCMA @8KHz, sendrecv, peer=- > ?? ? RX pt=8, stat last update: 00h:00m:02.427s ago > ?? ? ? ? ?total 1pkt 0B (40B +IP hdr) @avg=0bps/4bps > ?? ? ? ? ?pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) > ?? ? ? ? ? ? ? ? ? (msec)? ? min ? ? avg ? ? max ? ? last? ? dev > ?? ? ? ? ?loss period: ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 > ?? ? ? ? ?jitter ? ? : ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 > ?? ? TX pt=8, ptime=20ms, stat last update: never > ?? ? ? ? ?total 3.5Kpkt 570.2KB (712.8KB +IP hdr) @avg 63.9Kbps/79.8Kbps > ?? ? ? ? ?pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > ?? ? ? ? ? ? ? ? ?(msec)? ? min ? ? avg ? ? max ? ? last? ? dev > ?? ? ? ? ?loss period: ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 > ?? ? ? ? ?jitter ? ? : ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 > ?? ? RTT msec ? ? ? : ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 > > But I don't know what exactly I can do here, I have tried to disable VAD I > dont think NAT could be the problem, Im using 3G and I have tried in several > Wi-Fi to test if it was a router problem, besides that it works fine in the > same networks with pjsip 1.0 and the same SIP servers accounts. > By the other hand, when I try to use the --use-stun1 or --no-vad from > command line in pjsua it always says "Invalid input" don't know why? > > Getting Around NAT Explained above in checklist #5 > > Check that the Call is Connected to the Sound Device in the Conference > Bridge. When i use cl to see the port list I see: > > Conference ports: > Port #00[ 8KHz/20ms/1] ? ? iPhone IO device? transmitting to: #3 > Port #01[ 8KHz/20ms/1] ? ? ? ? ? ? ringback? transmitting to: > Port #02[ 8KHz/20ms/1] ? ? ? ? ? ? ? ? ring? transmitting to: > Port #03[ 8KHz/20ms/1] sip:?349201031 at abc.domain.com? transmitting to: #0 > > So, apparently ports are connected properly ant transmitting. > > Check that CPU Utilization is not Too High This is not the case, is the only > app running in the iPhone. > > I?tried to put all the information I could to help everybody to understand > better my problem and I can add more information if needed/requested. > I even tried to add multitasking and running on background support to pjsip > 1.0 but even I was able to receive calls on background and use other apps > while in a call with or without speaker,?whenever the apps get locked I stop > receiving calls and when I tried to return to my app I receive a SIGPIPE > error that I haven't been enable to fix. > I really think?that making work the pjsip 1.8 is a better approach and I > feel I'm just missing an easy step to make it work, so I hope someone here > could take the time to help me. > By the way: I also been able to receive incoming calls on pjsua from pjsip > 1.8, I dont know if this is related with the fact that Im not receiving rtp > packets when on call, and this is another problem I can try to fix by > miself, just wanted to note in case is related with this problem. > Thanks in advance for your time! > Regards, > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
