[asterisk-dev] Problems with RTCP

[John Lange]

I've been exploring the possibility of doing something useful with the
RTCP statistics that Asterisk (and more specifically chan_sip) generate
and I've encountered a number of issues.

After enabling the RTCP stats at the CLI as well as capturing them from
the RTPAUDIOQOS channel variable I noticed that the numbers it was
reporting didn't make sense so I did a packet capture and analyzed it
with Wireshark.

In the scenario where we have:

phone <-> asterisk <-> Cisco sip gateway (PRI)

a call placed from the phone to the outside will generate 3 RTP
streams. 

The first stream is the inbound call progress (the ringing sound) coming
from the PRI gateway and then, once the destination answers, a new pair
of RTP streams is created for in and outbound audio.

The problem appears to be that only the statistics for the first stream
are kept and all other statistics are discarded.

I can certainly understand how keeping statistics for mult-leg SIP calls
would be problematic so I'm wondering if it makes sense to keep all the
stats and present them all at the end as one big string? Alternatively
they could all be totalled ?

And a minor point, the values for ssrc in the RTPAUDIOQOS variable are
presented as decimal values but they are actually a hex number in the
packet streams. This makes comparing logs from Asterisk to packet
captures in Wireshark very difficult.

Before I do any digging around in the code I'm just wondering if anyone
has any comments on the current status of RTCP in Asterisk?

Has it been improved much in 1.6 for example?


John Lange
www.johnlange.ca