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

[Olle E Johansson]

24 apr 2007 kl. 03.40 skrev 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.
>

I was working on a manager update with this information, propably #3  
in your list.

/O