[OpenSIPS-Users] Audio calls not working on 3G
jayesh1017 at gmail.com
Fri Mar 7 13:08:55 CET 2014
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.
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?
> Users mailing list
> Users at lists.opensips.org
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