asterisk users - Dec 2007 - Call Quality Issues With 2 Trixbox's - Router Issue?

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
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How are the calls being transferred from Box A to Box B?

 

On what box is the receptionist registered too?

 

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel
Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?

 

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.

 

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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

What codec are you using?

PaulH


On Wed, 2007-12-12 at 13:00 +1100, 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
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G729 All Around.

Daniel Cole  (CCNA)
Technical Support
[http://www.hugonet.com.au/clients/hugonet.gif]
Ph: 1800 424 683
Fax: 03 5221 7659
e: dcole at hugonet.com.au<mailto:dcole at hugonet.com.au>
w: hugonet.com.au<http://www.hugonet.com.au/>

-----------------------------------------------------------------------------------

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________________________________
From: Paul Hales [mailto:phales at asteriskit.com.au]
Sent: Wednesday, 12 December 2007 4:10 PM
To: Daniel Cole
Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router
Issue?


What codec are you using?

PaulH


On Wed, 2007-12-12 at 13:00 +1100, 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

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Hi Paul,

Where abouts exactly is the best place to get these figures from?

I have been checking iax2 show netstats, which does give some figures. These
appear not to be accurate though, as when there are multiple inter-site calls,
the result for one channel of audio can show no jitter or latency, but another
will have some jitter and latency. Or is this a weird way for the problem to
show its head?

Thanks,

Daniel Cole  (CCNA)
 P Please consider the environment before you print this e-mail or any
attachments.


________________________________
From: Paul Hales [mailto:phales at asteriskit.com.au]
Sent: Wednesday, 12 December 2007 4:40 PM
To: Daniel Cole
Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router
Issue?


Hmmm......wierd....

Are you getting an weird jitter/latency figures in the CLI?

PaulH


On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote:
G729 All Around.
Daniel Cole  (CCNA)

 P Please consider the environment before you print this e-mail or any
attachments.



________________________________

From: Paul Hales [mailto:phales at asteriskit.com.au]
Sent: Wednesday, 12 December 2007 4:10 PM
To: Daniel Cole
Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router
Issue?




What codec are you using?

PaulH


On Wed, 2007-12-12 at 13:00 +1100, 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

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