[Asterisk-Dev] RT(C)P-debugging

[Karl Brose]

I think this (voip quality monitoring)  is an area of growing interest.


Filip Olsson wrote:

>Hi guys,
>
>I've posted a patch in #2570 for doing _some_ more verbose
>RTP-debugging.
>It adds the CLI commands 'rtp debug [ip host[:port]]' and 'rtp no
>debug'.
>It works pretty much the same way sip debug does.
>
>Here's some output:
>Got RTP packet from 10.0.0.37:8000 (type 0, seq 4041, ts 1771635040, len
>160)
>Sent RTP packet to 10.0.0.38:8002 (type 0, seq 8584, ts 88800, len 160)
>
>Here you can see the payload type(more or less the codec), sequence
>number, timestamp and the size of the packet.
>
>It's handy to have when trying to figure out where those RFC2833 packets
>are going and why it's not working.
>
>Please test the patch and give me feedback on additional features you'd
>like to see in it or anything else such as that you would like to
>resolve the PT in the output. Put warnings up when we see a jump in
>sequence numbers? This would(not always) indicate a packet loss. Output
>the RFC2833 digit?
>
>The RTCP-version of this one is coming up soon. It's very nice to be
>able to get some information on the transmission quality such as number
>of lost packets and interarrival jitter.
>I'm planning on doing some kind of 'monitor' for RTCP RRs and SRs and
>doing some statistical calculations on them and log this for all/some
>calls. I think this would be _very_ useful when hunting down echos and
>other network problems in general. Please note that Asterisk don't send
>any RTCP-packets, some RTP-stacks send them to us. At the moment we
>don't do anything about them, but there's alot of useful information in
>them.
>I actually don't know if it would be useful for us to be sending SRs and
>RRs to our peers, will it take any kind of action depending on what we
>send? Can someone tell me if there's a RTP-stack that would change
>behaviour because of content in RTCP-packets?
>The only(?) reason I see where it would be useful to generate
>RTCP-packets is when the remote peer is another Asterisk(or other
>RTP-stack) that can do something useful with the packets(for monitoring
>purposes or controlling bandwidth).
>
>It would be cool to output all this information to monitor(generate
>fancy graphs?) call quality over time and identify individual hosts that
>have problems.
>
>What say you?
>
>//Filip
>
>
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