Dear Daniel,
My whole set-up remains same in both the cases, only change is running
RTPproxy with kamailio server and without running RTPproxy server.
Now When i run RTPproxy with kamailio server, RTP packets has to reach up to
RTPproxy server to get relay to other end client right ?
But how this
Hi,
When you don't use RTPproxy, RTP traffic is sent end-to-end.
What is the RTP path in case of client-to-client RTP ?
Is it the same ?
Daniel
On 02/25/2014 06:34 PM, Ravi wrote:
> Dear Daniel,
>
> Thank you for the response,
>
>> Do you have any router, firewall something between clients
Dear Daniel,
Thank you for the response,
>Do you have any router, firewall something between clients and mediaproxy ?
Ya I have Routers And switch in between clients and Rtpproxy, My set-up is
as follows:
SIP Clients<-->WAP+WiMax CPE <--> WiMax Base station <>
RTPproxy+Kamailio se
Hi Ravi,
if you media/rtp proxy is receiving such packet loss, it means that
something behind him is cutting the traffic off, somehow.
So you should investigate there, in their configuration.
If your analysis is right, it seems your loosing around 80% of your packets.
Do you have any router, firew
Dear Daniel,
Thank you again,
Ya i am investigating on this issue, by the way can you give any comments on
my questions in the previous mail? I just wanna clarify those things to
rectify this packet loss issue. i googled about those questions but still
ended with the same confusion status and did
Hi,
Yes packet loss is very high.
Please investigate around that.
Daniel
Ravi wrote:
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>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>sr-users@lists.sip-router.org
>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr
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
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 analy
Dear Frank,
thank you for the response.
>Are you trying to run on virtualization? You might be having CPU
contention issues.
No im not running on Virtualisation. My set-up is as mentioned in previous
mail.
please help me in resolving this issues.
Regards,
Ravi
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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 call
On Feb 21, 2014, at 4:24 AM, Daniel Grotti wrote:
> Hi,
> it looks like your platform/network is introducing jitter in RTP packets.
> Mediaproxy/rtpproxy usual introduce a very low jitter but it has no
> impact to performances at all.
>
> It's hard to say, you should investigate in your platfor
Hi,
it looks like your platform/network is introducing jitter in RTP packets.
Mediaproxy/rtpproxy usual introduce a very low jitter but it has no
impact to performances at all.
It's hard to say, you should investigate in your platform first, and
then in your network devices.
For example, when are
Dear All,
Anybody has any clue about this issue.
Please help me.
Regards,
Ravi
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Sent from the Users mailing list archive at Nabble.com.
Dear Kamailio'ns,
I have configured kamailio (V 4.0) with RTPproxy server. Audio/Video call
establishments are just fine.
With this i have some unknown behaviour in my set-up is that When i am
running kamailio server with RTPproxy instance, i am experiencing Jitter,
Latency, pixelled audio/video c
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