[Freeswitch-users] Support for RTCP report via SIP PUBLISH

[Helmut Kuper]

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Hello Patrick,

> Thanks. I just noticed that there is another patch in Jira that also
> relates to RTCP reports: http://jira.freeswitch.org/browse/FS-949
> If it makes sense perhaps you and Guillaume can work together to merge
> your patches and come up with one?

I just emailed Guillaume. His patch seems to add some more informations
from periodically received RTCP reports to RECV_RTCP_MESSAGE while mine
is collecting the Vendor specific RTCP report after a call has terminated.

But basically I guess we both just want a detailed QOS report to keep
track of the network/voice quality.


> Also it's probably a good idea to ask the core developers what their
> intentions are regarding RTCP and see if both your patches fit their
ideas.

That was exactly my intention in my first email yesterday.
"I wonder whether this feature is planed in near future."

The biggest pain in my project is currently that I have nearly no tools
to debug RTP data for calls from the past. Mainly I have have rare cases
where the first 0.5 second of "Hello" is missed and also rare cases
where one RTP datastream is completely missed. Sure, this could be
caused by Network or SIP-Devices instead by FS, but back in reality I am
the one who has to deliver the first facts for starting debugging
outside of FS.


best regards
helmut
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