Never mind my last remark. The sampling rate has nothing to do with it. Bill, can you tell me how to make calls within the same lan? On 2015-03-05 15:06, David Desopper wrote: > 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 > > > > _______________________________________________ > 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... 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