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
