Hi Saul, Which endpoint(s) experienced distorted audio? Just FYI, G722 codec should also use 8kHz for RTP timestamp, however we experienced that some UAs still use 16kHz for G722 RTP timestamp, and perhaps this was the case. Btw, we've tried to make stream.c a bit smart by detecting remote RTP timestamp and readjusting local RTP timestamp, but somehow it could be wrong. So perhaps you could check the RTP timestamp in the RTP packets traffic manually, and please report back to us :) BR, nanang On Wed, Dec 2, 2009 at 7:25 AM, Jens Jorgensen <jbj1 at ultraemail.net> wrote: > You are really "talking" at 16kHz. Please see > http://tools.ietf.org/html/rfc3551#page-14 for an explanation. To > paraphrase: the RTP clock rate for G.722 was incorrectly specified in an > old RFC as 8000. However since it has been around a long time the RFC > has officially /kept/ this error around for compatibility reasons. Cute huh? > > As to the distortion problems, unfortunately I have no ideas for you. :-( > > Saul Ibarra Corretge wrote: >> Hi! >> >> I was testing g722 codec and I'm facing some distortion when trying the ZipDX demo (sip:wbdemo at conf.zipdx.com) I've gone though the bug tracked and found this issue regarding the clock rate (http://trac.pjsip.org/repos/ticket/486) however PJSUA is telling me that I'm using 16KHz: >> >> Intive is correct: >> INVITE sip:wbdemo at conf.zipdx.com SIP/2.0 >> Via: SIP/2.0/UDP 10.10.10.2:5060;rport;branch=z9hG4bKPjbNXyvh9iXDA8cdVGfW63Wq8QEh2zMQUQ >> Max-Forwards: 70 >> From: <sip:10.10.10.2>;tag=sk0bfd-EK92bJwKVasEXH1PKLYOKN6sO >> To: sip:wbdemo at conf.zipdx.com >> Contact: <sip:10.10.10.2:5060> >> Call-ID: 2KdBxLWweGZkzXu4b5Vjv6tESipzzgMq >> CSeq: 18006 INVITE >> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS >> Supported: replaces, 100rel, timer, norefersub >> Session-Expires: 1800 >> Min-SE: 90 >> User-Agent: PJSUA v1.4-trunk/i386-apple-darwin10.2.0 >> Content-Type: application/sdp >> Content-Length: ? 453 >> >> v=0 >> o=- 3468657952 3468657952 IN IP4 10.10.10.2 >> s=pjmedia >> c=IN IP4 10.10.10.2 >> t=0 0 >> a=X-nat:0 >> m=audio 4002 RTP/AVP 103 102 104 113 3 0 8 9 101 >> a=rtcp:4003 IN IP4 10.10.10.2 >> a=rtpmap:103 speex/16000 >> a=rtpmap:102 speex/8000 >> a=rtpmap:104 speex/32000 >> a=rtpmap:113 iLBC/8000 >> a=fmtp:113 mode=30 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:9 G722/8000 >> a=sendrecv >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 200OK also: >> SIP/2.0 200 Ok >> Via: SIP/2.0/UDP 10.10.10.2:5060;received=81.204.182.64;branch=z9hG4bKPjbNXyvh9iXDA8cdVGfW63Wq8QEh2zMQUQ;rport=12528 >> From: <sip:10.10.10.2>;tag=sk0bfd-EK92bJwKVasEXH1PKLYOKN6sO >> Call-ID: 2KdBxLWweGZkzXu4b5Vjv6tESipzzgMq >> CSeq: 18006 INVITE >> To: sip:wbdemo at conf.zipdx.com;tag=telStage-39d294b6-4b1506a0 >> Content-Length: 205 >> Content-Type: application/sdp >> Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS, INFO, REGISTER, SUBSCRIBE, MESSAGE >> Session-Expires: 1800;refresher=uas >> Contact: <sip:76.74.151.123;transport=udp> >> Server: ZipDX-3.10.4 >> Supported: timer >> >> v=0 >> o=telStage 1781 3468657952 IN IP4 76.74.151.123 >> s=- >> c=IN IP4 76.74.151.123 >> t=0 0 >> m=audio 12596 RTP/AVP 9 101 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> But when I see the call quality (dq in pjsua) I see that audio is in 16KHz: >> >>>>> dq >>>>> >> ?13:06:44.445 ? ?pjsua_app.c >> ? [CONFIRMED] To: sip:wbdemo at conf.zipdx.com;tag=telStage-39d294b6-4b1506a0 >> ? ? Call time: 00h:00m:52s, 1st res in 322 ms, conn in 328ms >> ? ? SRTP status: Not active Crypto-suite: (null) >> ? ? #0 G722 @16KHz, sendrecv, peer=76.74.151.123:12596 >> ? ? ? ?RX pt=9, stat last update: 00h:00m:00.068s ago >> ? ? ? ? ? total 2.6Kpkt 416.1KB (520.2KB +IP hdr) @avg=64.0Kbps/80.0Kbps >> ? ? ? ? ? pkt loss=0 (0.0%), discrd=1 (0.0%), dup=0 (0.0%), reord=1 (0.0%) >> ? ? ? ? ? ? ? ? (msec) ? ?min ? ? avg ? ? max ? ? last ? ?dev >> ? ? ? ? ? loss period: ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 >> ? ? ? ? ? jitter ? ? : ? 0.000 ? 2.863 ?70.625 ? 5.375 ? 2.654 >> ? ? ? ?TX pt=9, ptime=20ms, stat last update: 00h:00m:11.333s ago >> ? ? ? ? ? total 1.3Kpkt 222.2KB (277.8KB +IP hdr) @avg 34.1Kbps/42.7Kbps >> ? ? ? ? ? pkt loss=10 (0.7%), dup=0 (0.0%), reorder=0 (0.0%) >> ? ? ? ? ? ? ? ? (msec) ? ?min ? ? avg ? ? max ? ? last ? ?dev >> ? ? ? ? ? loss period: 200.000 200.000 200.000 200.000 ? 0.000 >> ? ? ? ? ? jitter ? ? : ? 2.375 ? 2.500 ? 2.625 ? 2.500 ? 0.088 >> ? ? ? RTT msec ? ? ? : ? 0.000 ? 0.000 ? 0.000 ? 0.000 ? 0.000 >> >> Is it just a "printing bug" or am I really talking at 16KHz? Any clue of what could be causing that distortion? >> >> Let me know if I can provide more information on this. >> >> >> Thanks in advance, >> >> > > > -- > Jens B. Jorgensen > jbj1 at ultraemail.net > > > _______________________________________________ > 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 >
