asterisk users - Oct 2005 - delays with IAX2 and Meetme

Hi there

 

I am using IAX2 softphones dialing into meetme conferences. I also have
jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
having is that as soon as there is a delay from a participant, then the
delay continues until the participant hangs up and dials in again. When
dialing in again the delay seems to go.

 

It seems to me as though as soon as the server registers a delay from a
participant, then it causes delay on all further packets from that
participant.

 

Does anyone have any ideas what the problem could be?

 

Many thanks

 

Steven

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In article <E1EPcnp-0002ek-IH@piglet.barn.za.net>,
Steven Langley <steven@intellinc.co.za> wrote:
> 
> I am using IAX2 softphones dialing into meetme conferences. I also have
> jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
> having is that as soon as there is a delay from a participant, then the
> delay continues until the participant hangs up and dials in again. When
> dialing in again the delay seems to go.
> 
> It seems to me as though as soon as the server registers a delay from a
> participant, then it causes delay on all further packets from that
> participant.
> 
> Does anyone have any ideas what the problem could be?
Yes, there are a few possibilities. Firstly, are you using ztdummy for
timing? Which kernel version? If 2.6, have you ensured that USE_RTC is
correctly defined in ztdummy.c?

Look in bugs.digium.com at bug IDs 3599 and 4252 - they might be relevant.

Yesterday I found another mechanism which could give rise to both a delay
and broken audio - I found it with OH323 channels, but it might possibly
arise on other channel types too. It concerns a backlog building up in
the channel driver and never being drained by meetme because of blocking
in the pseudo-device when trying to write the contents of a large frame.

In app_meetme.c, try replacing this:

 careful_write(fd, f->data, f->datalen);

with this:

 if (f->datalen <= CONF_SIZE) {
   careful_write(fd, f->data, f->datalen);
 } else {
   ast_log(LOG_WARNING, "Discarding large frame (%d bytes) from channel
%s\n", f->datalen, chan->name);
 }

and see if it helps.

I haven't yet submitted the above change to mantis.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

Hi Tony

 

Thanks for the reply and for posting the code. I added the code and
recompiled Asterisk, but unfortunately it did not resolve the issue. It
basically trapped all of the incoming audio and wrote to the error log
instead of outputting it. So basically it never seemed to go to the
careful_write statement.

 

To answer your questions: Firstly, I am using kernel 2.6, but am not using
Ztdummy. I am using a digium card for timing. I have run a test on it, and
it seems to be working properly.

 

I am using a client built using IaxClient, and am now looking at the
possibility that the delay might be a client issue instead of a server
issue. What is the best tool to use to run tests on my server and clients to
narrow down the source of the delay?

 

Many thanks

 

Steven

 
> Date: Wed, 12 Oct 2005 10:41:33 +0000 (UTC)
> From: tony@softins.clara.co.uk (Tony Mountifield)
> Subject: [Asterisk-Users] Re: delays with IAX2 and Meetme
> To: asterisk-users@lists.digium.com
> Message-ID: <diip8t$dq8$1@softins.clara.co.uk>
> 
> In article <E1EPcnp-0002ek-IH@piglet.barn.za.net>,
> Steven Langley <steven@intellinc.co.za> wrote:
>> 
>> I am using IAX2 softphones dialing into meetme conferences. I also have
>> jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
>> having is that as soon as there is a delay from a participant, then the
>> delay continues until the participant hangs up and dials in again. When
>> dialing in again the delay seems to go.
>> 
>> It seems to me as though as soon as the server registers a delay from a
>> participant, then it causes delay on all further packets from that
>> participant.
>> 
>> Does anyone have any ideas what the problem could be?
> 
> Yes, there are a few possibilities. Firstly, are you using ztdummy for
> timing? Which kernel version? If 2.6, have you ensured that USE_RTC is
> correctly defined in ztdummy.c?
> 
> Look in bugs.digium.com at bug IDs 3599 and 4252 - they might be relevant.
> 
> Yesterday I found another mechanism which could give rise to both a delay
> and broken audio - I found it with OH323 channels, but it might possibly
> arise on other channel types too. It concerns a backlog building up in
> the channel driver and never being drained by meetme because of blocking
> in the pseudo-device when trying to write the contents of a large frame.
> 
> In app_meetme.c, try replacing this:
> 
> careful_write(fd, f->data, f->datalen);
> 
> with this:
> 
> if (f->datalen <= CONF_SIZE) {
>   careful_write(fd, f->data, f->datalen);
> } else {
>   ast_log(LOG_WARNING, "Discarding large frame (%d bytes) from channel
%s\n", f->datalen, chan->name);
> }
> 
> and see if it helps.
> 
> I haven't yet submitted the above change to mantis.
> 
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: tony@softins.co.uk - http://www.softins.co.uk
<http://www.softins.co.uk/> 
> Play: tony@mountifield.org - http://tony.mountifield.org
<http://tony.mountifield.org/> 

 

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Hi

 

Thanks for the reply

 

I do actually use the |q option to disable the enter/exit sounds.

 

Steven

 
> Message: 15
> Date: Thu, 27 Oct 2005 10:25:32 -0500
> From: "Eric \"ManxPower\" Wieling"
<eric@fnords.org>
> Subject: Re: [Asterisk-Users] Re: delays with IAX2 and Meetme
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>     <asterisk-users@lists.digium.com>
> Message-ID: <4360F16C.4040800@fnords.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
 
> Steven Langley wrote:
>> Hi Tony
>> 
>>  
>> 
>> Thanks for the reply and for posting the code. I added the code and
>> recompiled Asterisk, but unfortunately it did not resolve the issue. It
>> basically trapped all of the incoming audio and wrote to the error log
>> instead of outputting it. So basically it never seemed to go to the
>> careful_write statement.
>> 
>>  
>> 
>> To answer your questions: Firstly, I am using kernel 2.6, but am not
using
>> Ztdummy. I am using a digium card for timing. I have run a test on it,
and
>> it seems to be working properly.
>> 
>>  
>> 
>> I am using a client built using IaxClient, and am now looking at the
>> possibility that the delay might be a client issue instead of a server
>> issue. What is the best tool to use to run tests on my server and
clients
to
>> narrow down the source of the delay?
> 
> 
>I'm pretty sure this is a known issue with enabling meetme enter/exit 
>sounds.  Turn them off (q option to MeetMe, I think)
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On Thu, Oct 27, 2005 at 02:10:51PM -0400, Dave Grey exclaimed:
>
>These appear to be a common problems, but after spending half a day  
>reading the wiki and list archives I have not gained much useful  
>knowledge beyond the fact that these are a common problems.  I am  
>hoping for some suggestions or pointers to further info.
>
>I have an ivr in my incoming context that does a background() and...  
>well, it is an ivr, no need to explain that, I guess.
>
>So, testing locally, it works wonderfully.  Testing through my DID,  
>provided by IPKall, it is decidedly hit-or-miss.  The digits seem to  
>be either not recognized at all or recognized incorrectly better than  
>half the time. Most often, I get the "invalid extension" playback
>that I have assigned to the i,1 exten.  For a while, I had two test  
>extensions, one 2000 and one 2001.  Dialing 2001 usually sent me to  
>2000 instead. What is making it hard for me to debug is that it  
>*sometimes* works, recognizing the extension I dialed correctly.
>
>My peer entry in sip.conf for IPKall contains dtmfmode=rfc2833 as per  
>their recommendation.  I have tried setting relaxedtmf=yes in the  
>general section, with no noticeable change.  I turned it off again,  
>since the problem seems to be too much relaxation in any case.   
>Looking at the console, I dial 7056 and it sees 7055, I dial 7056  
>again and it sees 75, I dial 7056 a third time and it sees 7055556,  
>etc.  Seems random and all over the place.  Packet loss and/or  
>ordering?  Aside from the dtmf issue, incoming calls on the DID work  
>fine and sound excellent.
>
This is a known issue that is fixed in CVS HEAD. Search for my prior
emails to track down the bug numbers. Unfortunately this is going to
require the upstream providers to upgrade to truly fix the issue.
Basically the RTP packets are coming out of order and have the wrong
sequence numbers. I started on a band-aid solution using ip_queue, but I
have not had time to finish it up. I will post here if it ever
materializes.

-Ryan

On Sun, 2005-10-30 at 23:42 -0500, Dave Grey wrote:
> On Oct 30, 2005, at 11:21 PM, trixter aka Bret McDanel wrote:
> >  I am starting a provider that gives free service
> 
> Nice! You went from listing free providers to becoming one. ;-)
> 
I had been planning on offering free service for a while, started to get
a little bit of time available and was able to put together what I had
already :)

> I would love an invite if you could use another tester.
> 
> Either way, good luck with this and thanks for providing another free  
> option.
It should be open to everyone soon.  Right now the Sacramento Asterisk
Users Group is the test ground, and I gotta move everything before I
open it up anyway, I dont have the bandwidth on the sip/iax2 server
right now to support hundreds of people.  I also plan on adding more
servers, right now I have one, a friend is donating some bandwidth and a
server for a second, and I am in talks with a few people in different
countries to donate some space for additional.


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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