On Sat, Dec 5, 2009 at 2:48 AM, Nanang Izzuddin <nanang at pjsip.org> wrote: > Which endpoint(s) experienced distorted audio? Hi all, I'm very interested in g722 in clients, especially for OS X. We run the VoIP Users Conference every Friday and it uses the ZipDX.com wideband conference bridge. When I'm home, I can use Windows XP and eyebeam to connect with good sound, but on the road on the Macbook, Counterpath hasn't put g722 in OS X yet. I have tried every OS X SIP client I have heard about and they all use pjsip. Every one of them has distortion problems. We know about the issue with the speci 16/8 khz etc, but no one has made it work properly. When I initiate a SIP call, some clients start out ok, but degrade quickly to distortion and choppiness. On Saul's client Blink, it sounds like the audio level I am hearing on my side is over-driven, not jitter or packet loss. Lowering the local audio radically seems to help. Maybe this is an OS X issue? Is there any way we could get someone to test on the ZipDX bridge some time? The owner of ZipDX, David Frankel is technical enough and has the tools to be able to help figure out what's happening, I believe. I personally will help test any way I can, as I do have a compelling motive to get this working. I can make recordings if need be. Please let me know if we can help and if there's way to get this fixed, as none of the current clients are usable at all in g722. Regards, Randy http://VoIPUsersConference.org > 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 >> > > _______________________________________________ > 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 >
