Pseudo ICE feature request for GTalk XMPP interop

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Hi Aaron,

we're probably not interested in adding GTalk's "ICE" protocol, since
we can't talk directly with it anyway. But it should be fairly simple
to create a media transport adapter for this (similar to SRTP; see
[1]), and add STUN send/receive capability into it (see STUN Session
in [2]). Then plug your adapter on top of existing UDP media transport
using https://trac.pjsip.org/repos/ticket/1173.

[1] http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm
[2] http://www.pjsip.org/pjnath/docs/html/group__PJNATH__STUN.htm

?Benny


On Fri, Dec 10, 2010 at 6:56 PM, Aaron Clauson <aaron at sipsorcery.com> wrote:
> I'm running a SIP server and have been investigating integrating with
> Google's GTalk service over XMPP. The integration turned out to be fairly
> painless and I can translate between SIP and XMPP to handle call signalling.
> The problem is the ICE format Google use to set up the media.
>
> Google are a little bit behind with their ICE implementation and instead of
> utilising RFC5389 STUN requests and the sequence of them specified in the
> ICE RFC 5245 specification they rely on RFC 3489 STUN requests. Google's
> STUN request flow is actually very simple, the client needs to send a STUN
> (RFC3489) binding request to which the Google media server will respond and
> then send its own STUN binding request to which the client must respond. So
> that's one STUN binding request each from the client and server. After that
> RTP starts flowing exactly the same as for a standard SIP call. The
> specification detailing the Google approach is
> http://code.google.com/apis/talk/call_signaling.html.
>
> The reason I'm sending this email is to see if any pjsip developers are
> interested in adding the ability for the pjsip stack to set up media with
> the Google GTalk service by adding the ability to send the STUN requests and
> responses it requires. AS far as I know there is no current client SIP/media
> stack that can interoperate with Google's GTalk service (as far as B2BUA's
> go Asterisk and FreeSWITCH can but I'm after a UA).
>
> Regards,
> Aaron
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>



[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux