Hi,<br>Yes packet loss is very high.<br>Please investigate around that.<br><br>Daniel<br><br><br>Ravi <wingsravi777@gmail.com> wrote:<br><br><div dir="ltr"><div>Dear Daniel,<br><br>thank you very much for the reply,<br><br>As you suggested i did sip/rtp traffic analysis using tcpdump and wireshark captures.<br>Using Wireshark, i tried to analyse RTP packet loss and it is as follows:<br>
1) Client A to RTPproxy<br> Total RTP packets Sent: 1776 , RTP packets lost : 1060 (59.68%), Mean Jitter: 25ms<br><br>2) RTPproxy to Client B<br> Total RTP packets Sent: 1776 , RTP packets lost : 1060 (59.68%), Mean Jitter: 25ms<br>
<br><br> client A ------59.68%------> RTPproxy ------59.68%------> Client B<br><br>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 ?<br>
And with this i have one more question that :(Sorry if it is silly question, I am newbie in this)<br>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 ?<br>
<br>And i have attached Tcpdump based SIP captures for your better understanding.<br>Also find my Kamailio config file and please suggest me about anything can be done on script level to fine tune this issue ?<br><br></div>
Please help me in resolving this issue.<br><div><br>Awaiting your reply,<br><br>Regards,<br>Ravi<br><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Feb 21, 2014 at 11:23 PM, Daniel Grotti-4 [via SIP Router] <span dir="ltr"><<a href="/user/SendEmail.jtp?type=node&node=125289&i=0" target="_top" rel="nofollow" link="external">[hidden email]</a>></span> wrote:<br>
<blockquote style='border-left:2px solid #CCCCCC;padding:0 1em' class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="">
Hi Ravi,
<br>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.
<br><br>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.
<br>In particular, analyzing RTP traffic you will be able to see jitter value between Client A->rtpproxy and rtpproxy->Client B.
<br>So you can check if the traffic is already jittered or not.
<br>If not, it means that your server is adding jitter.
<br><br><br>Daniel
<br><br><br><br></div><div class="">On Friday, February 21, 2014 18:36 CET, Ravi <<a href="http://user/SendEmail.jtp?type=node&node=125260&i=0" rel="nofollow" link="external" target="_blank">[hidden email]</a>> wrote:
<br>
</div><div><div class=""><div class='shrinkable-quote'><br>> Dear Daniel,
<br>>
<br>> Thank you for the reply,
<br>>
<br>> What you are saying is right, but my problem with this set-up is, without
<br>> running rtpproxy server instance, only with running kamailio server
<br>> everything (audio/video) is just go fine. But when i start RTPproxy server
<br>> to achieve NAT traversal, audio/video calls are going badly with pixelled
<br>> video and latency, voice break kind of issues with audio.
<br>>
<br>> And my system set-up is like this :
<br>> Runing both Kamailio and RTPproxy in same machine on ubuntu (12.04)
<br>> platform. And i am working on Intranet infrastructure, so both the Rtpproxy
<br>> server and kamailio listening on Private IP address.
<br>>
<br>> Can you please tel me how can i check jitter levels and RTP packet loss
<br>> before reaching RTPproxy server in my network ?
<br>> Anything can be done on RTPproxy server ?
<br>>
<br>> Please help me in resoloving this issues.As i am new to this kind of
<br>> networking concepts.
<br>>
<br>> Awaiting replies.
<br>>
<br>> Regards,
<br>> Ravi
<br>>
<br>>
<br>>
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