Bill, I now believe that this is a audio hardware issue instead of something else. To get my USB headset working, I have to set the clock-rate at 48kHz, can this be causing the problem in your opinion? This would shift the problem from PJSIP to Raspberry Pi hardware. On 2015-03-04 10:36, David Desopper wrote: > Bill, > > I normally did use a turn server in both endpoints. I will try your > solution and get back to you as soon as I did. > > Thank you, > David > > Bill Gardner schreef op 3/03/2015 om 22:29: >> Hi David, >> >> OK. I see the call is connecting via a turn relay. Did you set up the >> ubuntu endpoint to use turn? It's surprising that ice didn't >> negotiate using the local RTP addresses on your LAN. Actually I can >> see that ice tests the local connection and it succeeds: >> >> 15:09:57.513 icetp00 ICE negotiation success after 0s:201 >> 15:09:57.515 icetp00 Comp 1: sending from host candidate 192.168.1.124:4015 to host candidate 192.168.1.110:4033 >> 15:09:57.516 icetp00 Comp 2: sending from host candidate 192.168.1.124:4001 to host candidate 192.168.1.110:4024 >> But then rasberry proceeds to use the media relay which seems wrong >> to me. I don't understand ice well enough to parse the issue from the >> logs. >> >> One suggestion is to drop all the registrar and ice/turn/stun options >> at both endpoints and just make the call between the LAN addresses, >> that should work and will verify you have working endpoints. Then you >> can go through the registrar and enable ice, which should negotiate >> the local RTP addresses. Then add turn relay to one or both >> endpoints. It should still use the local RTP addresses I would think. >> >> Regards, >> >> Bill >> >> On 3/3/2015 3:35 PM, David Desopper wrote: >>> 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 >>> >>> >>> >>> _______________________________________________ >>> 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 > > > > _______________________________________________ > 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/20150305/9094ad36/attachment.html>
