[pjsip] Distorted audio with g722

Nanang Izzuddin nanang at pjsip.org
Sun Dec 6 21:58:55 EST 2009


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
>




More information about the pjsip mailing list