Wireshark-dev: Re: [Wireshark-dev] Question regarding RTCP protocol
From: Alejandro Vaquero <[email protected]>
Date: Wed, 15 Nov 2006 12:32:59 -0800
Hi Andreina,
The RTCP has the filed SSRC that match the RTP stream that is referring to. So you can filter for example:
rtcp.ssrc.identifier == 1492336106 or rtp.ssrc == 1492336106

and that will display the RTP and RTCP messages of that stream with SSRC 1492336106 .
   Other way to identify the two is using the UDP port. According to 
the spect, the RTCP port is always the next odd port of the RTP stream 
UDP port. For example if RTP is using 8000, then the RTCP messages will 
use 8001. This is not always the case lately with all the NAT firewalls 
that are around, where not always the outside ports are translated in seq.
Alejandro

Andreina Toro wrote:
Hi everyone, I have a question regarding RTCP protocol. I have to measure the jitter, packet loss and roundtrip delay for each call. I know that in the statistics menu in the RTP Stream Window I can read the Mean Jitter and Packet Loss per call, and if in the general window I filter the RTCP protocol I´m able to read the roundtrip delay per packet per call in the sender´s report. My problem is that I can have a lot of calls between the same 2 elements, (between 2 IP´s addresses (A and B)), so in the RTP Stream Window I can read a lot of calls between 2 IP´s source and destination (A and B). How can I relate the roundtrip calculation value in the general window when I filter wiht RTCP protocol with a specific call in the RTP Stream Window if the info I have is the IP address for the source and destination (A and B) and there are a lot of calls between them. My question is how I can I know from which call is that RTCP sender report? Is there a specific field in the RTCP packet where I can relate that report with a specific call? So I can know for each call between 2 elements the 3 parameters (jitter, packet loss and roundtrip delay) Thanks for your time, looking forward for some help!
Andreina
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