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

[Jayesh Nambiar]

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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>
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