Bill, I am sorry for my numerous e-mails, but this problem is seriously bugging me because it is part of my master thesis, deadlines are creeping in fast. When I make a LAN call I get this information: Press a to answer or h to reject call a Answer with code (100-699) (empty to cancel): 200 14:40:39.123 pjsua_call.c !Answering call 0: code=200 14:40:39.126 pjsua_media.c ...Call 0: updating media.. 14:40:39.127 pjsua_aud.c ....Audio channel update.. 14:40:39.131 strm0x47bb44 .....Encoder stream started 14:40:39.132 strm0x47bb44 .....Decoder stream started 14:40:39.140 pjsua_media.c ....Audio updated, stream #0: PCMU (sendrecv) 14:40:39.141 pjsua_app.c ...Call 0 media 0 [type=audio], status is Active 14:40:39.142 pjsua_aud.c ...Conf disconnect: 2 -x- 0 14:40:39.144 conference.c ....Port 2 (ring) stop transmitting to port 0 (Hercules HD Exchange: USB Audio (hw:1,0)) 14:40:39.146 pjsua_aud.c ...Conf connect: 3 --> 0 14:40:39.147 conference.c ....Port 3 (sip:192.168.1.108) transmitting to port 0 (Hercules HD Exchange: USB Audio (hw:1,0)) 14:40:39.149 pjsua_aud.c ...Conf connect: 0 --> 3 14:40:39.151 conference.c ....Port 0 (Hercules HD Exchange: USB Audio (hw:1,0)) transmitting to port 3 (sip:192.168.1.108) 14:40:39.154 Master/sound !Underflow, buf_cnt=0, will generate 1 frame 14:40:39.153 pjsua_core.c ....TX 889 bytes Response msg 200/INVITE/cseq=12740 (tdta0x467ad8) to UDP 192.168.1.108:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.108:5060;rport=5060;received=192.168.1.108;branch=z9hG4bKPj9ErfYX2FuKbD1gEhQCuTMR7bhUmUYzm7 Call-ID: TR.3BVSkf3YKwlNsrT1D6-P0ffDN.r3X From: <sip:192.168.1.108>;tag=FJ6rb9ese.ANHOoZ9.CpfWsZxAZxcLzj To: <sip:192.168.1.124>;tag=mAL2aev6FkoPxy0h2PEHNm6Oi5CaZW90 CSeq: 12740 INVITE Contact: <sip:192.168.1.124:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 274 v=0 o=- 3634555234 3634555235 IN IP4 192.168.1.124 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 0 96 c=IN IP4 192.168.1.124 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.1.124 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 14:40:39.161 pjsua_app.c .......Call 0 state changed to CONNECTING >>> 14:40:39.169 pjsua_core.c .RX 352 bytes Request msg ACK/cseq=12740 (rdata0x45c2bc) from UDP 192.168.1.108:5060: ACK sip:192.168.1.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.108:5060;rport;branch=z9hG4bKPjTBaR1hYODm1YOsSjR9ZXhlQzY211Rw0u Max-Forwards: 70 From: <sip:192.168.1.108>;tag=FJ6rb9ese.ANHOoZ9.CpfWsZxAZxcLzj To: sip:192.168.1.124;tag=mAL2aev6FkoPxy0h2PEHNm6Oi5CaZW90 Call-ID: TR.3BVSkf3YKwlNsrT1D6-P0ffDN.r3X CSeq: 12740 ACK Content-Length: 0 --end msg-- 14:40:39.171 ec0x46e880 !Underflow, buf_cnt=0, will generate 1 frame 14:40:39.176 pjsua_app.c ...Call 0 state changed to CONFIRMED 14:40:39.243 ec0x46e880 !Underflow, buf_cnt=0, will generate 1 frame 14:40:39.374 Master/sound Underflow, buf_cnt=0, will generate 1 frame 14:40:39.431 Master/sound Underflow, buf_cnt=0, will generate 1 frame 14:40:39.488 Master/sound Underflow, buf_cnt=0, will generate 1 frame 14:40:39.544 Master/sound Underflow, buf_cnt=0, will generate 1 frame 14:40:39.592 Master/sound Underflow, buf_cnt=0, will generate 1 frame Can this be the cause? On 2015-03-05 15:24, David Desopper wrote: > 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 > > > > _______________________________________________ > 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/de9111ad/attachment.html>
