Bill, Thanks for your effort. I try to make calls between a pjsua client on a raspberry on one side and a pjsua client on an ubuntu system inside the same lan. I ran through the same steps on both end sytems. It is puzzeling me for a while now and I don't have a doubt in my mind that it's just me that's doing something stupidly wrong. Bill Gardner schreef op 3/03/2015 om 21:28: > 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 > > > > _______________________________________________ > 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/db869615/attachment.html>
