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 -------------- next part -------------- An HTML attachment was scrubbed... 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