Hi, The Randi's words 'over-driven audio level' sounds related to ticket #658 [1]. -- [1] http://trac.pjsip.org/repos/ticket/658 -- BR, nanang On Sat, Dec 5, 2009 at 4:38 PM, Randy R <randulo2008 at gmail.com> wrote: > 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 >> > > _______________________________________________ > 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 >
