asterisk users - Oct 2012 - Odd cracking with SIP->DAHDI

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?

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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