Hi David, The call stats show 8 packets sent and 449 received during an 8 second call. Given that you've checked the sound device using cc 0 0 it's puzzling that you don't hear any audio during calls. Are you sure the other endpoint is functional? One suggestion is to first make point to point calls between pjsip clients on your LAN before trying to connect to external clients. Another idea is to use wireshark to capture the RTP stream. The log looks OK to me, although I would expect more packets to be sent with vad off. Regards, Bill On 3/3/2015 10:18 AM, David Desopper wrote: > Hi all, > > For my master thesis I'm trying to use the pjsip libraries for > incorperated inside my own software project. But I can't seem to be > able to get pjsua working. Something clearly is going wrong, but I > have no idea what the problem seems to be. If I check the sound > quality paramaters during a call, I can see that I am not transmitting > any sound packets, but also, I can't hear any sound that clearly is > received. > > I did all the steps mentioned here: > https://trac.pjsip.org/repos/wiki/Audio_Problems/Getting_Around_Nat > but without succes. > > I also checked that my audio device is working by doing the command cc > 0 0 to echo my microphone input sound to my output. > > Pjsua is initialized like this: > > ./pjsua-arm-unknown-linux-gnueabihf --clock-rate=48000 --capture-dev=0 > --playback-dev=0 --use-ice --id=sip:XXX at sip.antisip.com > --registrar=sip:sip.antisip.com --realm=* --username=XXX > --password=*** --stun-srv=stun.pjsip.org --no-vad > > I think that the most important output is this: > > 15:10:06.492 pjsua_app_comm ! > [CONFIRMED] To: > <sip:XXX at sip.antisip.com>;tag=kiGDpg9qCdBHKMLgabtKNN1RR-6zUtNP > Call time: 00h:00m:08s, 1st res in 7505 ms, conn in 7779ms > #0 audio PCMU @8kHz, sendrecv, peer=91.121.78.130:51526 > SRTP status: Not active Crypto-suite: > ICE role: Controlled, state: Negotiation Success, comp_cnt: 2 > [0]: L:192.168.1.124:4015 (h) --> R:192.168.1.110:4033 (h) > [1]: L:192.168.1.124:4001 (h) --> R:192.168.1.110:4024 (h) > RX pt=0, last update:00h:00m:08.808s ago > total 449pkt 71.8KB (89.8KB +IP hdr) @avg=62.8Kbps/78.5Kbps > 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 2.247 9.250 0.750 1.971 > TX pt=0, ptime=20, last update:00h:00m:04.429s ago > total 8pkt 1.2KB (1.6KB +IP hdr) @avg=1.1Kbps/1.3Kbps > 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 : 4.425 4.425 4.425 4.425 0.000 > > > I included the full log of the call in attachment. > > Thanks for your help > > > _______________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150303/c3cd0c2c/attachment.html>
