[asterisk-users] Measuring voice quality with Asterisk

[Klaus Darilion]

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.
>