Hello;
I am trying to rtp processing with rtpproxy. i couldnt find
why it truncate the sdp body. there is no clue in debug mod 4.
Could you hava a look what is wrong?
rtpproxy running like that "rtpproxy -l public_ip -s
udp:127.0.0.1:7723 -F -m 1
Hi Safdar,
Il 05/11/15 06:17, Safdar Khan ha scritto:
Hi Daniel,
Your suggestion helped me a lot.Thanks.
But now i am running both rtp proxy and kamailio on same computer(i.e.
localhost) and according to syslog there is no error.
Further will you tell me how can i test that they are working tog
Subject: Re: [SR-Users] RTP Proxy to integrate with kamailio
Hi Daniel,
Your suggestion helped me a lot.Thanks.
But now i am running both rtp proxy and kamailio on same computer(i.e.
localhost) and according to syslog there is no error.
Further will you tell me how can i test that they are
Hi Daniel,
Your suggestion helped me a lot.Thanks.
But now i am running both rtp proxy and kamailio on same computer(i.e.
localhost) and according to syslog there is no error.
Further will you tell me how can i test that they are working together or
not, like in wireshark or any other method to tra
Hello,
you need to configure rtpproxy to listen on udp:192.168.5.97:7722 for
control commands, not on 127.0.0.1:7722. The 127.0.0.1 network interface
is available on inside same system, not from a remote system.
Cheers,
Daniel
On 04/11/15 13:41, Safdar Khan wrote:
> Hi friends,
> Its been two da
Hi friends,
Its been two days and still finding a way to install rtp proxy and
integrate it with kamailio.
I have checked running status of rtp proxy by *netstat* command in this
way...
*#netstat -pln | grep rttpproxy*
and i got the output like that(i assume its working):
*udp 0 0 127.0.0.1:
Hello,
thanks for sharing! Shouldn't the end of the loop (the 'done') be more
at the end of the script in order to handle all the saves streams, or is
it intended to run for a single saved rtp session?
Cheers,
Daniel
On 22/07/15 16:18, Alberto Sagredo wrote:
> Using RTPBreak and sox im able to c
Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later
you could convert to mp3 if liked
Just to share with list
https://github.com/albersag/rtpproxy-utils
Any comment is well appreciated
Best Regards
___
SIP Express Router (SER) a
Hi,
it looks like you call
> rtpproxy_manage("rwie");
twice.
In this case, you see the IP beeing added twice.
Kind regards,
Carsten
2013/10/24 Mahmoud Ramadan Ali :
> Hello ,
> This is the first time to put a question on sr-users mailing list and i
> hope to help me to solve my issue...i'm fol
Daniel, I got the answer for gdbthanks.
Arun
From: arun Jayaprakash
To: "mico...@gmail.com" ; Kamailio (SER) - Users Mailing
List
Sent: Thursday, July 11, 2013 6:03 AM
Subject: Re: [SR-Users] RTP proxy high CPU utilization...
Thank you D
Thank you Daniel. Can you let me know what do you mean by "attach with gdb",
thanks again.
Regards,
Arun
From: Daniel-Constantin Mierla
To: arun Jayaprakash ; Kamailio (SER) - Users
Mailing List
Sent: Thursday, July 11, 2013 5:57 AM
Subject: Re:
On 7/10/13 5:52 PM, arun Jayaprakash wrote:
I am still having issues with this, can someone shed some light on
this so I will know where to start looking for the problem as I am
new to Kamailio. Thank you.
Have you seen the reply:
http://lists.sip-router.org/pipermail/sr-users/2013-July/0788
I am still having issues with this, can someone shed some light on this so I
will know where to start looking for the problem as I am new to Kamailio.
Thank you.
Regards,
Arun
From: arun Jayaprakash
To: Kamailio Mailing List
Sent: Monday, July 8, 2013 10
Hello,
On 7/8/13 5:49 PM, arun Jayaprakash wrote:
Hello,
I have set up my Asterisk server with Kamailio where Kamailio handles
the user authentication as per
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
It is working well, I am able to make calls, two way sound
Hello,
I have set up my Asterisk server with Kamailio where Kamailio handles the user
authentication as per
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. It
is working well, I am able to make calls, two way sound, etc. The problem I am
facing is that after maki
Hello,
communication with rtpproxy is done from modules/rtpproxy module -- look
inside that module sources.
Only RTP port is received to update the sdp, RTCP is just increasing by 1.
Cheers,
Daniel
On 12/14/12 1:11 PM, Austin Einter wrote:
Dear All
I am looking at Kamailio's module, file, f
Dear All
I am looking at Kamailio's module, file, function that receives rtp and
rtcp port from rtp proxy.
Can somebody please let me know which file/function I need to check.
Thanks
Austin
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
Are you sure that force_rtp_proxy is not called?
Instead of
> if (nat_uac_test("8")) {
> *force_rtp_proxy();*
> } else {
> force_rtp_proxy();
> }
try:
xlog("L_ERR","route(4): method = $rm");
if (is_method("BYE|CANCE
Dear All,
I'm using the below config plan for routing my calls...The issue that
forcing rtp proxy is not working well and the rtp proxy is never
forcedCan you please provide me a guidance here?
if($rU=~"^00.*" )
{
if(!cr_route("default", "domain.com", "$rU",
The example that is in the link is using
rtpproxy_offer/rtpproxy_answer (don't know which files you were
browsing).
The script is provided as an example. Once you understand how it
works, you can build on top of it.
You need to invoke rtpproxy_offer for the first SIP request or reply
that is carr
Thanks for the references to examples. I've been through them, gotten
rid of force_rtp_proxy() since it's considered depreciated - but still
get the same error:
/usr/local/sbin/kamailio[31035]: ERROR:nathelper:force_rtp_proxy:
incorrect port 0 in reply from rtp proxy
I do this in my main route
Check out the examples provided in the code:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=modules_k/rtpproxy/examples/4to6.cfg;hb=ad7f00d840082989132f335914aa0db223a0e46e
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=modules_k/rtpproxy/examples
I've been looking around for documentation to understand rtp proxy
with Kamailio better, but with no luck (or I might be quite
thick-headed!).
Anyone know of example configurations with rtp proxy and Kamailio, so
I can start seeing what is wrong with my config (below)?
Thanks!
//Anders
On Wed,
Sorry, I didn't understand that. Is it good or bad that it is
asymmetrical? And if not, what should it be? As far as I understand
from my config, it first checks if NAT is set, and if so, it goes to
force_rtp_proxy if it's INVITE and unforce_rtp_proxy if it's a BYE. Is
that not correct?
> On
This suggests asymmetrical invocation of rtpproxy, for example in 200
OK but not corresponding INVITE.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On
Hi,
I'm trying to get the RTP proxy to work with Kamailio (1.5), but right
now I am getting this error here:
Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]:
ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp
proxy
There is probably something wrong in my cfg - for the use
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