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, -- Saul Ibarra Corretge AG Projects
