jay bing schrieb: > hi, > > thanks for the replay, > > again, why does I need implementation of srtp in the server side? don't > I do srtp agaist the peer I am taking to? or the srtp is agaist the server? Not with Asterisk. Asterisk is a PBX which decodes RTP and transform it into the internal format. In some cases (direct-bridging in chan_sip, or canreinvite=yes) you might have luck as then ASterisk acts just as relay or media will be "reinvited" to be directly end-to-end. regards klaus > > > > > On Tue, Oct 27, 2009 at 9:27 AM, Sa?l Ibarra <saghul at gmail.com > <mailto:saghul at gmail.com>> wrote: > > Not at the moment. Yu'll need to wait untill SRTP is fully implemented > in Asterisk: https://issues.asterisk.org/view.php?id=5413 > > On Mon, Oct 26, 2009 at 10:10 PM, jay bing <jaya.bing at gmail.com > <mailto:jaya.bing at gmail.com>> wrote: > > hi, > > why do I need support of the sip server to run SRTP? > > the asterisk doesn't just relay the rtp packets? > > can it be configured to just relay rtp packets? > > thanks > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org <http://blog.pjsip.org/> > > > > pjsip mailing list > > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > -- > /Sa?l > http://www.saghul.net <http://www.saghul.net/> | > http://www.sipdoc.net <http://www.sipdoc.net/> > > _______________________________________________ > Visit our blog: http://blog.pjsip.org <http://blog.pjsip.org/> > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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
