[SIPForum-discussion] RTCP Call Quality Monitoring

[Nikos Leontsinis]

Rtcp has many limitations as a protocol.
1.It doesn't measure one way delay (which is needed for mos)  so you
measure round trip and you falsely assume that your network is
symmetric.
2. You don't know what you measure since you don't know who is sending
the rtcp report back to you
the service provider or the carrier? I have the carrier environment
experience in mind.
3. Not everyone implements rfc 3550.
4.It is based on sampling as a control protocol hence it is not accurate.
You might like to have a look at the http://www.voipfuture.com/ an rtp
based solution but you still have the limitation that you only measure
your own backbone not the calls end to end accurately as you should.
In my opinion this is a well overlooked topic across the industry.


/nikos

Nikos Leontsinis
skype: leontsinis2


On 4 June 2014 00:27, Jorge Montesino <jmmo1981 at gmail.com> wrote:
> Dear All,
>
> I would like to know if you can recommend me an opensource platform or any
> other good for the price that i could use to gather RTCP packets information
> in order to monitor voip call quality. We do not proxy the calls (no RTP
> will be gather by the platform) in order to optimized bandwidth usage. We
> plan to do this installing the recommended platform in a standalone server
> and then configure a restricted SSH user into the media servers and capture
> the information by TCPDUMP permanent filter.
>
> Please let me know if you have any recommendation.
>
> Thanks a lot
>
> Jorge
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>