Dear Daniel,
thank you very much for the reply,
As you suggested i did sip/rtp traffic analysis using tcpdump and wireshark
captures.
Using Wireshark, i tried to analyse RTP packet loss and it is as follows:
1) Client A to RTPproxy
Total RTP packets Sent: 1776 , RTP packets lost : 1060 (59.68%), Mean
Jitter: 25ms
2) RTPproxy to Client B
Total RTP packets Sent: 1776 , RTP packets lost : 1060 (59.68%), Mean
Jitter: 25ms
client A ------59.68%------> RTPproxy ------59.68%------> Client B
It seems like RTP packets are dropped before they reached RTPproxy server,
With this i should have to look into my hardware between Client A to
RTPproxy right ?
And with this i have one more question that :(Sorry if it is silly
question, I am newbie in this)
How that RTP packets sent and lost ratios are same before and after
RTPproxy server ? i mean If Packets are dropping before reaching RTPproxy
,then RTPproxy has suppose to forward the packets as much it recieves (i.e
Recieved packets= Packets sent - Packets lost). but then how this before
and after RTPproxy Packets ratio is same ?
And i have attached Tcpdump based SIP captures for your better
understanding.
Also find my Kamailio config file and please suggest me about anything can
be done on script level to fine tune this issue ?
Please help me in resolving this issue.
Awaiting your reply,
Regards,
Ravi
On Fri, Feb 21, 2014 at 11:23 PM, Daniel Grotti-4 [via SIP Router] <
ml-node+s1086192n125260...@n5.nabble.com> wrote:
> 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 <[hidden
> email]<http://user/SendEmail.jtp?type=node&node=125260&i=0>>
> 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
> >
> >
> >
> > --
> > View this message in context:
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> > Sent from the Users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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kamailio.cfg (43K)
<http://sip-router.1086192.n5.nabble.com/attachment/125289/0/kamailio.cfg>
videocallWithrtpproxy.pcap (54K)
<http://sip-router.1086192.n5.nabble.com/attachment/125289/1/videocallWithrtpproxy.pcap>
videocallWithoutrtpproxy.pcap (31K)
<http://sip-router.1086192.n5.nabble.com/attachment/125289/2/videocallWithoutrtpproxy.pcap>
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