For the curios, with the RTCP patches applied against rtp.c for
Asterisk 1.2.0b2, you get the following new goodies:
CLI> help rtp rtcp
rtp rtcp debug Enable RTCP debugging
rtp rtcp debug ip Enable RTCP debugging on IP
rtp rtcp no debug Disable RTCP debugging
Turning on rtcp debugging, you get in progress RTCP reports looking like:
----
Sent RTCP SR to 192.168.1.121:16471
Our SSRC: 408251416
Sent(NTP): 1130992373.3367305216
Sent(RTP): 200160
Sent packets: 1251
Sent octets: 200160
Report block:
Fraction lost: 0
Cumulative loss: 0
IA jitter: 0.0005
Their last SR: 0
DLSR: 4.1580 (sec)
Sending RTCP RR to 192.168.1.121:16471
Our SSRC: 408251416
Their SSRC: 1069050633
Fraction lost: 0
Cumulative loss: 0
IA jitter: 0.0005
Their last SR: 0
DLSR: 4.1980 (sec)
----
Upon call termination, you get a report:
------
RTP-stats
* Our Receiver:
SSRC: 1069050633
Received packets: 964
Lost packets: 0
Jitter: 0.0004
Transit: -0.0014
RR-count: 5
* Our Sender:
SSRC: 408251416
Sent packets: 1447
Lost packets: 0
Jitter: 0
RTT: 0.000000
-----
Cheers,
Simon P. Ditner
On 11/2/05, Simon P. Ditner <[EMAIL PROTECTED]> wrote:
> After some very dry reading (RFC3261 - SIP, RFC1889 - RTP), I found
> some interesting (ha!) information in section 6.3 of RFC1889 on RTCP
> Sender/Receiver Reports -- with some good keywords in hand, I googled
> through the asterisk mailing lists to discover that this feature is
> coming Real Soon Now(TM), since 2004:
>
> http://bugs.digium.com/view.php?id=2863
>
> So, now I'm reading through the rather lengthily feature report, and
> seeing how far along things are. There's a fairly substantial patch
> attached to the bug report for rtp.c that looks like it will provide
> debug output for RTCP. With luck, I can shoe-horn that into Asterisk
> 1.0.9, without moving to -HEAD.
>
> Cheers,
> Simon P. Ditner
>
