[SIPForum-discussion] STUN, TURN and ICE

[John Downing]

Hi Manish:

You need to examine your network architecture carefully before considering
implementation of STUN. As a minimum you need to determine if there is a
NAT device associated with their of the subnets you identified.  Along with
NAT there are any number of additional reasons your RTP stream my be
interrupted between the two endpoints.

On Wed, Mar 6, 2013 at 2:26 PM, Manish Thakkar <enggmanish at gmail.com> wrote:

> Hi,
>
> Currently while being involved in webRTC calls, we have a concern with
> calls from Browser to Browser (B2B). If caller and callee are not part of
> the same sub net then there is no audio path established (Signalling is
> fine).
>
> I believe use STUN, TURN servers would help overcome the issue.
>
> Can someone please provide more info on the use of the servers and how to
> implement them in a test scenario.
>
> I appreciate your help, in advance
>
> Thanks,
> Manish
>
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-- 
Thanks... John

John Downing
John at TrainingCity.com
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