asterisk users - Aug 2009 - Measuring voice quality with Asterisk

Hi!

I want to use Asterisk as load generator to test quality degradation 
with increased load (e.g. testing other SIP equipment or IP-links).

Is anybody aware of such a setup with Asterisk - is it possible to get 
RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)?

Thanks
Klaus

On 27/08/09 9:24 PM, Klaus Darilion wrote:
> I want to use Asterisk as load generator to test quality degradation
> with increased load (e.g. testing other SIP equipment or IP-links).
>
> Is anybody aware of such a setup with Asterisk - is it possible to get
> RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)?
Looks like Tzafrir is building a branch that may interest you:

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=213876
Log:
A branch for an RTP streaming backend for res_monitor (1.4)

A new branch to add channel monitoring to a remote server as RTP
streams. The recordings are intended to be sent to an Oreka/Orex server.

Metadata about the RTP stream is sent in dummy SIP INVITE and BYE
messages.

This branch includes the code developed vs. 1.4 as this is the code that
is actually tested. A branch based on trunk will be available soon.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

Hi Matt!

Matt Riddell schrieb:
> On 27/08/09 9:24 PM, Klaus Darilion wrote:
>> I want to use Asterisk as load generator to test quality degradation
>> with increased load (e.g. testing other SIP equipment or IP-links).
>>
>> Is anybody aware of such a setup with Asterisk - is it possible to get
>> RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering
...)?
> 
> Looks like Tzafrir is building a branch that may interest you:
> 
> URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=213876
> Log:
> A branch for an RTP streaming backend for res_monitor (1.4)
Would be great if svn-view would be opened for public again - e.g. using 
HTTP authentication using the Mantis accounts.
> A new branch to add channel monitoring to a remote server as RTP
> streams. The recordings are intended to be sent to an Oreka/Orex server.
> 
> Metadata about the RTP stream is sent in dummy SIP INVITE and BYE
> messages.
Is this somewhere documented?

thanks
klaus


> 
> This branch includes the code developed vs. 1.4 as this is the code that
> is actually tested. A branch based on trunk will be available soon.
>

27 aug 2009 kl. 11.24 skrev Klaus Darilion:
> Hi!
>
> I want to use Asterisk as load generator to test quality degradation
> with increased load (e.g. testing other SIP equipment or IP-links).
>
> Is anybody aware of such a setup with Asterisk - is it possible to get
> RTP statistics out of Asterisk (e.g. jitter, packet loss,  
> reordering ...)?
>
Check the RTPAUDIOQOS variable documented in channelvariables.txt :-)

You propably want to catch it in the "h" extension and put it in the  
CDRs or something.

/O

Olle E. Johansson schrieb:
> 27 aug 2009 kl. 11.24 skrev Klaus Darilion:
> 
>> Hi!
>>
>> I want to use Asterisk as load generator to test quality degradation
>> with increased load (e.g. testing other SIP equipment or IP-links).
>>
>> Is anybody aware of such a setup with Asterisk - is it possible to get
>> RTP statistics out of Asterisk (e.g. jitter, packet loss,  
>> reordering ...)?
>>
> 
> Check the RTPAUDIOQOS variable documented in channelvariables.txt :-)
> 
> You propably want to catch it in the "h" extension and put it in
the
> CDRs or something.
That is good for after-call statistics. I am currently trying to get 
real-time stats.

regards
klaus

On 31/08/09 8:47 PM, Klaus Darilion wrote:
>
>
> Olle E. Johansson schrieb:
>> 27 aug 2009 kl. 11.24 skrev Klaus Darilion:
>>
>>> Hi!
>>>
>>> I want to use Asterisk as load generator to test quality
degradation
>>> with increased load (e.g. testing other SIP equipment or IP-links).
>>>
>>> Is anybody aware of such a setup with Asterisk - is it possible to
get
>>> RTP statistics out of Asterisk (e.g. jitter, packet loss,
>>> reordering ...)?
>>>
>>
>> Check the RTPAUDIOQOS variable documented in channelvariables.txt :-)
>>
>> You propably want to catch it in the "h" extension and put it
in the
>> CDRs or something.
>
> That is good for after-call statistics. I am currently trying to get
> real-time stats.
Do the rtcp stats give out info over the manager?  Maybe you could use 
those?  They definitely come up in the console if enabled.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

Matt Riddell schrieb:
> On 31/08/09 8:47 PM, Klaus Darilion wrote:
>>
>> Olle E. Johansson schrieb:
>>> 27 aug 2009 kl. 11.24 skrev Klaus Darilion:
>>>
>>>> Hi!
>>>>
>>>> I want to use Asterisk as load generator to test quality
degradation
>>>> with increased load (e.g. testing other SIP equipment or
IP-links).
>>>>
>>>> Is anybody aware of such a setup with Asterisk - is it possible
to get
>>>> RTP statistics out of Asterisk (e.g. jitter, packet loss,
>>>> reordering ...)?
>>>>
>>> Check the RTPAUDIOQOS variable documented in channelvariables.txt
:-)
>>>
>>> You propably want to catch it in the "h" extension and
put it in the
>>> CDRs or something.
>> That is good for after-call statistics. I am currently trying to get
>> real-time stats.
> 
> Do the rtcp stats give out info over the manager?  Maybe you could use 
> those?  They definitely come up in the console if enabled.
Yes, those events are reported over AMI.

In my first tests I connected via AMI and fetched a channel list, and 
then for each channel I fetched CHANNEL(qos,audio,all). Of course that 
did not scaled. Thus, I wonder if reading RTP events scale.

Further I found out that the RTCP events do not have a reference to a 
channel. Thus, it is not possible to find out which channel 
received/sent this RTCP packet.

As the QoS settings are also stored within the rtp channel structure, I 
was thinking of adding some functionality to rtp handler to export QoS 
stats every few seconds and when the call ends.

regards
klaus
>