[asterisk-dev] Extensions to RTPAUDIOQOS for SIP/other channels

[John Todd]

I have a requirement to log the IP address of the actual RTP 
endpoints to which a set of Asterisk servers is communicating.  Due 
to the typical wide array of resellers, re-re-re-sellers, SRV 
records, redirects, and any other confusions it is not possible to 
accurately determine where the media flows based just on what is 
intended, and often I find myself in the situation where the 
vendor(s) claim that they can't help me to debug a particular problem 
unless I give them the IP address of the RTP speakers (SDP data) on 
both sides of the call and not just the SIP data.  It seems to me 
that this would be useful for any media stream that uses RTP, and not 
merely SIP.

I understand that during transfers and re-invites, this IP address of 
the RTP far-end speaker may change.  However, it seems that the 
RTPAUDIOQOS fields are at least a sane location in which to put the 
RTP stats for a particular stream, and let the higher layer 
signalling redirection argument happen at some later time (if this 
not already resolved.)

So, two questions:

  1) Does RTPAUDIOQOS currently change back to zero when 
transfers/re-invites occur?  (I could test this myself, I know, but 
it would require a lot more acrobatics and hopefully someone just 
knows.)

  2) Would anyone object to the addition of the "far-end" IP address 
for a given RTP stream being included in the RTCPAUDIOQOS string 
somewhere, or available otherwise within the dialplan?  If there is 
any consensus that this might be useful, we'll code up and/or pay for 
the patch to be submitted to the community.  This seems like a 
trivial patch.


JT