[pjsip] Passing SIP through TURN

Guilherme Balena Versiani guibv at nymgo.com
Thu Jun 7 09:39:56 EDT 2012


Long story short, you are wrong. My 2 cents back.
Em 07/06/2012 10:17, "Saúl Ibarra Corretgé" <saul at ag-projects.com> escreveu:

>
> On Jun 7, 2012, at 3:10 PM, Guilherme Balena Versiani wrote:
>
> > No, my application should be aimed to run even under hard network
> conditions.
> >
> > There are some network conditions which I need to multiplex both audio
> and signalling in the same channel, and I will use TURN in these cases.
> >
>
> In that case you might be trying to use the wrong protocol. My 2 cents.
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
> _______________________________________________
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120607/4c108c0b/attachment.html>


More information about the pjsip mailing list