[asterisk-users] Odd cracking with SIP->DAHDI

[Steve Totaro]

On Tue, Oct 16, 2012 at 1:31 PM, Richard Kenner <kenner at gnat.com> wrote:

> We recently set up a SIP trunk between an office in NY running Asterisk and
> an office in Paris (running Alcatel).  All works fine if a SIP phone on the
> NY system talks to the Paris PBX.  But if something on DAHDI (a PRI or
> MeetMe) talks to the Paris PBX, there's a low-volume crackling.  This isn't
> clipping because it also occurs when there's no legitimate sound.  It's
> sort of a mild version of what you used to get when a POTS pair had a
> ground short.  This occurs no matter what size originates the call.
>
> pings show round trip times of around 100ms, ranging from around 200 to 80
> ms.  Packet loss is zero.  The fact that SIP->SIP works fine suggests the
> issue isn't related to IP issues.
>
> I tried adding a jitter buffer, but that didn't make a difference.
>
> I've tried this sending just ULAW and G722 and allowing everything, but no
> difference.  The SDP that comes back from Paris doesn't list any audio
> codecs and is:
>
>     v=0
>     o=default 1350406175 1350406175 IN IP4 10.10.22.246
>     s=Asterisk PBX 10.7.1
>     c=IN IP4 10.10.22.246
>     t=0 0
>     m=audio 32000 RTP/AVP 0 101
>     a=sendrecv
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=ptime:20
>     a=maxptime:90
>     m=video 0 RTP/AVP 31 34 34 98 99 104
>     a=rtpmap:31 H261/90000
>     a=rtpmap:34 H263/90000
>     a=rtpmap:34 H263/90000
>     a=rtpmap:98 h263-1998/90000
>     a=rtpmap:99 H264/90000
>     a=rtpmap:104 MP4V-ES/90000
>     a=sendrecv
>
> Does anybody have any ideas as to what I should look at next?
>
>
cat proc/interrupts?

 http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards

Thanks,
Steve T
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