[asterisk-users] SIP & Echo

Tue May 22 09:09:29 MST 2007

Thanks guys for the tips. I will try that.

Alex

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony Francis
Sent: Tuesday, May 22, 2007 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP & Echo

Asterisk wrote:
>
> In Sip.conf I have the following: canreinvite=no
>
>  
>
> No, all telephones are on the same subnet, handled by the same switch. 
> I cannot verify if anything has been changed since I installed & 
> configured the network, but as far as I know the whole network 
> configuration is pretty straightforward, without any routing madness.
>
>  
>
> Kind Regards,
>
> Alex
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *David 
> Gomillion
> *Sent:* Tuesday, May 22, 2007 4:34 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] SIP & Echo
>
>  
>
> Are your phones reinviting? Do you have any strange routing weirdness, 
> or are they all on a single subnet?
>
> On 5/22/07, *Asterisk* < asterisk at abraxas.si 
> <mailto:asterisk at abraxas.si>> wrote:
>
> I tried with the ping ... all of the phones respond in ca. 0.3ms, so 
> network seems to be OK. More than 90% of CPU on * box is idle even in 
> peak times, so this shouldn't cause echoes either, right? Hmmm, so 
> handset could be an issue, but did anyone ever experience any handset 
> problems with Polycom IP SoundPoint 430 :-) ?
>
>  
>
> I will check the headsets and any possibilities of acoustical echo. 
> Besides that, if we rule out the network, and the CPU on the * box, is 
> there anything else that could be causing echoes on internal SIP calls?
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com> 
> [mailto:asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com>] *On Behalf Of *David 
> Gomillion
> *Sent:* Tuesday, May 22, 2007 3:22 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] SIP & Echo
>
>  
>
> We experience echo too from time to time. It's usually 
> headset-related, but not always. I ran a persistent ping on one of the 
> phones, and we diagnosed a wiring problem with it. Other phones needed 
> a new handset. The problem is that these problems need to be fixed on 
> the phone NOT hearing echo.
>
> On 5/22/07, *Asterisk* <asterisk at abraxas.si 
> <mailto:asterisk at abraxas.si>> wrote:
>
> How could I check if bandwith or/and latency is causing it?
>
> If I do SIP show peers it says OK (13 ms) for all peers. I guess there 
> is a way to gather more detailed info on SIP calls and latency?
>
> * box is connected to the 1Gb switch with 1Gb connection, and clients 
> have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP 
> hardphones connected to the * box.
>
> Thanks, Alex
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com> [mailto: 
> asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of 
> Alexandre VERNIOL
> Sent: Tuesday, May 22, 2007 2:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP & Echo
>
> Hi,
>
> Could be bandwith or/and latency ... Many causes...
>
>
> Alex
>
> Asterisk a écrit :
> > Hello all,
> >
> > One of our clients reported that they are experiencing echo in SIP calls
> > (even on internal ones). What do you think could be causing echo in
> > internal SIP calls?
> >
> > We're using Polycom telephones, do you think they could be causing it?
> >
> > Thanks,
> > Alex
> >
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watch the comm between the switch and a host with issues using tcpdump 
-Avvv host <host-ip>

This can give you an idea of what is going on, also if they are all in 
the same subnet, you are giving your ast box unnecessary strain forcing 
it to relay the RTSP stream, I would set that canreinvite to yes. For 
reference on sip and how re-invite works, please read 
http://www.faqs.org/rfcs/rfc2543.html
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