[asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss
Edoardo Serra
edoardo.serra at webrainstorm.it
Sat Mar 24 01:53:16 MST 2007
Hi Francois,
f6hqz-m at hamwlan.net ha scritto:
> Hi men,
>
> I have already encountered some issue like this with few switches (very
> known great brand) which doesn't like VoIP traffic !
I also have switches of a very known great brand !!
It was so strange to me that I didn't consider a network problem...
> Check by drectly connected the VoIP equipment - if you can - with
> temporary long Ethernet cables bypassing the tested switch to see what
> happens in this case.
I'd try to bypass the switch someway but every server neeeds to have its
own public ip address..
I'll put an RTP proxy somewhere...
> You can also tell to "qualify" with a longer delay, but this could not
> help in case of regulary frames losses.
What about turning qualify off ?
Do you think taht Asterisk is stopping RTP when it loose a qualify packet ?
Or is the RTP traffic itself that is lost by the switches ?
> Good luck !
It couldn't be more appropriate...
Tnx for help ;)
Edoardo
>
> Francois BERGERET,
> France.
>
> -----Message d'origine-----
> *De :* asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] *De la part de*
> Rajeev Natarajan
> *Envoyé :* samedi 24 mars 2007 08:14
> *À :* Asterisk Users Mailing List - Non-Commercial Discussion
> *Objet :* Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss
>
> Well, we have add similar issues - do you use a media gateway /.IP
> Phones / softphones as your extensions?
>
> We were running Audiocodes and for some reason (I suspect a poor
> ethernet switch), when there are more than 15 people using the line,
> Audiocodes will not respond to a qualify and asterisk will drop the
> call. Turned off qualify (removed qualify=yes) and <still keeping
> fingers crossed> things seem fine.
>
> Rajeev
>
> On 3/23/07, *Edoardo Serra* <edoardo.serra at webrainstorm.it
> <mailto:edoardo.serra at webrainstorm.it>> wrote:
>
> Hi all,
> I'm having a problem with some Asterisk servers
> interconnected with
> each other using IAX (I also tried with SIP without solving the
> problem)
>
> Sometimes, with apparently no reason, some peers become UNREACHABLE
> (I have qualify=yes in iax.conf) and REACHABLE again as soon as
> another qualify test is made.
>
> Our users are also complaining about audio loss during their calls,
> apparently randomly, everything goes ok for days and bad for
> another few
> days.
>
> I strongly believe the 2 problems are strictly related because
> in the
> logs I see REACHABLE / UNREACHABLE messages only for certains days
> without regularity.
> The days in wich i see a lot of messages are exactly the days with
> most of complaint about audio loss
>
> I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
> are quite always during business hours, this makes me think at
> somewhat
> related to load (cpu load, badwidth load, calls load, etc...)
>
> But, looking at hardware specs of our lan, servers and average
> load I
> don't think they are over-stressed.
>
> Our servers are all:
> 2 x Intel(R) Xeon(TM) CPU 3.20GHz
> 1 GB RAM
> 2 x IDE HDDs Software RAID 1
> Asterisk 1.2.13 with res_perl
> Gentoo Linux
> Some of them has a Sangoma card connected with an E1
>
> Most ot these are on the same LAN, interconnected with a 1 GB switch
> (I don't think it should be a bandwidth problem).
>
> Load averages of these server is varying from 0.5 to 1.0
> (I guess it should be ok)
>
> On each server we don't have more than 50 concurrent calls
> (bridged SIP <-> IAX2 or IAX2 <-> ZAP)
>
> Used codec is mostly G729
>
> Sometimes on asterisk cli i see some messages like
> "Avoided initial deadlock for '0x9fd130', 10 retries!"
> I don't know if it could be somehow related.
>
> Someone of you can point me in the right direction ?
>
> Tnx in advance
>
> Regards
>
> Ing. Edoardo Serra
> WeBRainstorm S.r.l.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
> <http://Easynews.com> --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list