[OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork
Rajesh Babu
rajesh.babu at goodcoresoft.com
Fri Sep 27 08:34:04 CEST 2013
Hi Mike,
This is log i am geting wheni try to start the service
[root at centos64 rtpproxy-1.2.0]# tailf /var/log/messages
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not
respond, disable it
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not
respond, disable it
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon process
exiting with 0
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 10:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
When you use the residential script almost all configuration come alredy
working for this
i have a tutorial (in portuguese ( i think that i should translate to
english :) )) , where you can see a routing script working with nat
http://opensips.com.br/wiki/index.php?title=Opensips_1.9
You can take a look at modules documentation too
http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html
There is on this maillist too a lot of discussions about this, below you can
see one case
http://opensips.org/pipermail/users/2011-January/016130.html
If you get some information from an old version of opensips probably will be
necessary to take a look on the module documentation to check about little
diferences , but i think that this is the start point :)
and if you is new to opensips i recommend to you the book about opensips (
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )
2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>
Hi Mike,
Thanks for the response, I am totally new to this world, can you please
help me by directing to on how to configure links. It will be great.
Thanks in advance
Regards
Rajesh
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 12:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
you should configure the nathelper and rtpproxy, this should help in you
issue.
2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network between two users i don't have any
issue, where as from outside the network, even though i can see the user
registered in my server i am not able to call registered user (I see the
user in my UL show listing). The call is established but i am not able to
talk (Mean the audio and video are not getting transffered).
Where as messages are going fine without any issue. I guess it is because
message transmit over XMPP where calls on SIP right.
I am really struck and i don't know how to proceed, please help me out
Thanks
Rajesh
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