[OpenSIPS-Users] Audio calls not working on 3G

[Rajesh Babu]

Hi Jayesh,

 

    Wifi network is not within my work network. I have disabled all the
firewall in my server. If firewall is blocking the packets, but I could
place the call using a another opensource app. 

 

-Thanks

Rajesh

 

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Jayesh Nambiar
Sent: Friday, 7 March, 2014 8:09 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Audio calls not working on 3G

 

Your work firewall must be blocking packets when you test on 3G. The Wifi
must be within your work network !! I hope you are using RTPProxy or
MediaProxy to handle media when originated from NATed clients. If yes, you
dont need STUN and TURN as of now.

 

--- Jayesh

 

On Fri, Mar 7, 2014 at 5:01 PM, Rajesh Babu <rajesh.babu at goodcoresoft.com>
wrote:

Hi All,

 

  I use Opensips 1.9.1 and have enabled RTP and Nating in the configuration,
Whenever I use to connect the calls using my 3G connection, call gets
connected but my voice is not being heard, whereas though wifi everything is
working fine. I tried connecting with Linphone I didn't face any issue,
where as whenever I try connecting using my app which on top of CSip I am
getting this issue. This issue is not getting replicated over wifi, I am
getting this issue only on 3G. My carrier is not blocking any packets from
my side as different opensource client is letting me make calls over the
SIP.

 

Some blogs stated that configuring Stun will solve this issue, I tried doing
it but no luck. In some other blog they where stating I can go with TURN
Server, I need to know whether Turn servers solve these issues and someone
can put me over the installation and using guide for the same.

 

Can someone please direct me on the right track please?

 

-Thanks

Rajesh


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