[Asterisk-Users] AArgh, * and the 7960
Iain Stevenson
iain at iainstevenson.com
Thu May 20 01:20:39 MST 2004
I have ethereal installed and I'll do a full call trace. The Catch 22 is I
don't have access to access to a source of repeatable (ie recorded) content
accessed through IAX. That would help in producing traces for the ATA and
7960 for comparison. I mainly use IAX for non-critical international
business calls to people who wouldn't want to be * testers.
Iain
--On Tuesday, May 18, 2004 7:22 pm -0600 "brian k. west" <brian at bkw.org>
wrote:
> Lets look at this and FIX the problem instead of hacking it. What you
> need to do is install etherreal and capture a call and parse the
> timestamp info to see if they are slipping. Because they are perfect
> here.
>
> bkw
>
> ----- Original Message -----
> From: "Brian Cuthie" <brian at systemix.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, May 18, 2004 5:07 PM
> Subject: Re: [Asterisk-Users] AArgh, * and the 7960
>
>
>>
>> Iain,
>>
>> This is a known issue with the Cisco phone and Asterisk having to do
>> with a change made later in the cvs tree. Try 1.0 stable, or modify
>> rtp.c to comment out the two lines as follows:
>>
>> /* Re-calculate last TS */
>> rtp->lastts = rtp->lastts + ms * 8;
>> // if (!f->delivery.tv_sec && !f->delivery.tv_usec) {
>> /* If this isn't an absolute delivery time,
>> Check if it is close to our prediction,
>> and if so, go with our prediction */
>> if (abs(rtp->lastts - pred) < 640)
>> rtp->lastts = pred;
>> else {
>> ast_log(LOG_DEBUG, "Difference is %d, ms
>> is %d\n", abs(rtp->lastts - pred), ms);
>> mark = 1;
>> }
>> // }
>> } else {
>>
>> This seems to work for me. Others may have more insight.
>>
>> -brian
>>
>>
>> Nik Martin wrote:
>>
>> > Out of context, this isn't much information. Is your network
>> > connection
> OK?
>> > Is your broadband provider having troubles? Has some upstream hardware
>> > changed that you may not be aware of?
>> >
>> >
>> >
>> >
>> >> -----Original Message-----
>> >> From: asterisk-users-admin at lists.digium.com
>> >> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
>> >> Iain Stevenson
>> >> Sent: Tuesday, May 18, 2004 1:29 PM
>> >> To: asterisk-users at lists.digium.com
>> >> Subject: [Asterisk-Users] AArgh, * and the 7960
>> >>
>> >>
>> >>
>> >> I've just had the most appalling performance from * ever. Dialling:
>> >>
>> >> Cisco 7960 => asterisk => IAX
>> >>
>> >> produces sound drop outs so extreme that the call is useless.
>> >> I noted this
>> >> in an earlier post. Dialling:
>> >>
>> >> Cisco ATA186 => asterisk => IAX
>> >>
>> >> is fine.
>> >>
>> >> Frankly, I think this is such a bad problem that it should be
>> >> sorted in
>> >> advance of any of the new features that seem to be getting
>> >> such prominence
>> >> nowadays. It was not present earlier in the year and I
>> >> haven't upgraded my
>> >> 7960. So I don't think you can point the finger entirely in Cisco's
>> >> direction.
>> >>
>> >> Iain
>> >> _______________________________________________
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>> >>
>> >>
>> >
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