17 Feb
2014
17 Feb
'14
11:12 p.m.
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 calls. But when i run only Kamailio server
without RTPproxy service running, audio/video calls are just fine (no any
jitter, latency kind of issues).
I also tried with Mediaproxy (AG projects) server but the results are same
as like in Kamailio/RTPproxy case.
What could be the problem with this set-up ? How can i solve this issue ?
Please anybody help me in resolving this problem.
Any help will greatly appreciate.
Regards.
Ravi.
18 Feb
18 Feb
9:25 p.m.
Dear All,
Anybody has any clue about this issue.
Please help me.
Regards,
Ravi
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21 Feb
21 Feb
2:24 p.m.
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 you using media-relay, do you receive packets
already jittered ? If yes problem could come from some network devices
in front of you mediaproxy.
Daniel
On 02/17/2014 07:12 PM, Wingsravi R wrote:
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 calls. But when i run only
Kamailio server without RTPproxy service running, audio/video calls are
just fine (no any jitter, latency kind of issues).
I also tried with Mediaproxy (AG projects) server but the results are
same as like in Kamailio/RTPproxy case.
What could be the problem with this set-up ? How can i solve this issue ?
Please anybody help me in resolving this problem.
Any help will greatly appreciate.
Regards.
Ravi.
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9:15 p.m.
On Feb 21, 2014, at 4:24 AM, Daniel Grotti <dgrotti(a)sipwise.com> 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 platform first, and
then in your network devices.
For example, when are you using media-relay, do you receive packets
already jittered ? If yes problem could come from some network devices
in front of you mediaproxy.
Are you trying to run on virtualization? You might be having CPU contention issues.
--FC
10:42 p.m.
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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10:36 p.m.
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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10:52 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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http://sip-router.1086192.n5.nabble.com/RTPProxy-Mediaproxy-issue-tp125121p…
Sent from the Users mailing list archive at Nabble.com.
_______________________________________________
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23 Feb
23 Feb
10:23 p.m.
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+s1086192n125260h50(a)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:
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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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:
http://sip-router.1086192.n5.nabble.com/RTPProxy-Mediaproxy-issue-tp125121p…
Sent from the Users mailing list archive at
Nabble.com.
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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7 comments
4 participants
participants (4)
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Daniel Grotti -
Frank Carmickle -
Ravi -
Wingsravi R