21 Feb
2014
21 Feb
'14
11:22 p.m.
Hi Ravi,
yes it means that when RTP traffic passes through your media-relay you have traffic, if
you don't use media-realy RTP traffic is end-to-end between clients.
To check jitter and other values you can capture your SIP/RTP traffic on your kamailio
server with "tcpdump" for example and analyze the call with wireshark.
In particular, analyzing RTP traffic you will be able to see jitter value between Client
A->rtpproxy and rtpproxy->Client B.
So you can check if the traffic is already jittered or not.
If not, it means that your server is adding jitter.
Daniel
On Friday, February 21, 2014 18:36 CET, Ravi <wingsravi777(a)gmail.com> wrote:
Dear Daniel,
Thank you for the reply,
What you are saying is right, but my problem with this set-up is, without
running rtpproxy server instance, only with running kamailio server
everything (audio/video) is just go fine. But when i start RTPproxy server
to achieve NAT traversal, audio/video calls are going badly with pixelled
video and latency, voice break kind of issues with audio.
And my system set-up is like this :
Runing both Kamailio and RTPproxy in same machine on ubuntu (12.04)
platform. And i am working on Intranet infrastructure, so both the Rtpproxy
server and kamailio listening on Private IP address.
Can you please tel me how can i check jitter levels and RTP packet loss
before reaching RTPproxy server in my network ?
Anything can be done on RTPproxy server ?
Please help me in resoloving this issues.As i am new to this kind of
networking concepts.
Awaiting replies.
Regards,
Ravi
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