[asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?

[Andres]

Do an RTP analysis with Wireshark of a sample call.   That could 
probably narrow down the source of the problem.  I would suspect you 
will either see some jitter or packets out of order.

Daniel Cole wrote:

> Hello Everyone,
>
> We have recently installed a pair of Trixbox servers in for a client 
> of our. They have two locations, with one server each. The servers 
> terminate 3 standard POTS lines into a Sangoma A200D card. The servers 
> are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon 
> Processors). We are using Trixbox 2.2, and G729 all around.
>
> Each site has two (2) 512k/512k ADSL connections terminating into a 
> Cisco 877W router (using an additional 'dumb' modem in a separate VLAN 
> for the extra dsl connection). Using policy based routing, all Voice 
> Data goes over one DSL connection (the one that terminates directly 
> into the router), and all other traffic (e.g. Web and VPN) goes out 
> the second connection (the bridged dumb dsl modem).
>
> We are also the ISP for this client, and as thus we have full 
> monitoring of our Layer 2 and Layer 3 networks. From our analysis, it 
> doesn't appear that there is any issue in these networks. We have 
> other customers using the VoIP service, who have not complained of 
> these issues.
>
> Now for the Fun part!
> The client is complaining of issues with inter-site calls. They are 
> reporting issues with crackly and broken speech, and horrible jitter 
> (or packet loss). This presents a huge issues, because they have one 
> receptionist answering all calls for both sites. So if a call comes in 
> from the other site, it automatically an inter-site call, and the 
> quality falls out of it. If the call is then transfered back to the 
> originating site, the audio 'bounces' between the two sites, which add 
> to the call quality degradation.
>
> We have been monitoring the router while these incidents have been 
> reported, and it does not appear to be a bandwidth issue. The DSL tail 
> used for Voice gets to no more then 120k in each direction (we have 
> tested the links, and can pull data at 53k/s between sites). CPU usage 
> floats at around 20-25% under load. The router has only shows major 
> packet loss (that we can tell) when REALLY pushing it in testing (e.g. 
> 10+ calls between sites).
> We have enabled the SIP jitter buffer, as well as the IAX jitter 
> buffer, which appeared to make a huge difference, but the issue is 
> still ongoing.
>
> These issues have also been reported with some outbound VoIP calls. 
> Internal calls, and calls directly in or out of the Sangoma card are 
> clear, with no issues reported.
>
> Does anyone have any thoughts on what could be causing these issues? 
> We have been racking our brains here, and have tried everything that 
> we can think of. These system is a million times better then what is 
> what when it was first installed, but it is still not where it should 
> be in terms of quality.
>
> Any thoughts/ideas are most welcome.
>
> Thank you
>
>  
>
> *Daniel Cole  **(CCNA)** *
>
> //
>
>
>  P Please consider the environment before you print this e-mail or any 
> attachments.
>  
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation Provided by http://www.api-digital.com--
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Andres
Technical Support
http://www.telesip.net