thanks, sorry, about "implement on an Asterisk server" I just mean that I'll be using RTP on Asterisk.
I've known the SRTP plug-in on Asterisk, but I also heard something that about SRTP being implemented on ekiga (at least here http://mail.gnome.org/archives/ekiga-devel-list/2006-October/msg00012.htmland this was about 4 years ago), maybe i ought to check the development list Thanks, On Wed, Nov 24, 2010 at 4:23 PM, Jānis Rukšāns <[email protected]>wrote: > On Wed, Nov 24, 2010 at 8:18 PM, Fedor vonBock <[email protected]> > wrote: > > I'm confused a bit with how ekiga can be integrated with any SIP > server. I > > just want to know how the Real TIme Protocol is involoved here (since > that > > is what I plan to implement on an Asterisk server) > > What do you mean with "plan to implement on an Asterisk server"? > Asterisk already "kinda" supports SIP - all you have to do is to > configure the clients and the dial plan. > > > I understand that SIP is mainly a negoatiation protocol while RTP is the > > protocol that carries the audio (or vidieo) and I plan to use Secure RTP > > later on. > > Well, technically SIP by itself is a signalisation protocol - it > carries around info about call setup, progress and stuff like that, > plus info about the media (audio and video). The SIP standard requires > that every SIP user agent [1] supports SDP (Session Description > Protocol) as the media description language, and an offer/answer model > for negotiating the actual media that will be used during the call > (defined in RFC 2543 [2]). The latter in turn requires that RTP must > be supported as a way to carry the media over the net. > > So this is how we get to "Ekiga can be integrated with any SIP server" > (in theory [3]). For the media endpoints, RTP is required due to the > above, and anything in the middle cares only about the SIP stuff. > Asterisk is like a "phone on steroids" and therefore qualifies as an > "endpoint" here, and thus is a subject to the RTP requirement. > > As for the SRTP, as far as I know, it is not supported by Ekiga; and > you need a patched Asterisk - it doesn't support SRTP out of the box. > > > If you plan to set up an Asterisk, I suggest you to read a bit on SIP > and associated protocols to get a glimpse on how they work together. > Even if not strictly required, that knowledge might save you some > headache later. > > Cheers > > -- > Ian > > [1] A SIP user agent is a piece of software/hardware that actually > processes the SIP messages, rather than just passing them around. As > SIP is peer-to-peer protocol, the term "server" is ambiguous here, as > most user agents are both clients (they send requests and process > responses) and servers (they process requests and respond to them). > This way, Ekiga is a "server", too. Compare this to HTTP, where a web > browser only sends requests, and a web server only responds to > requests. > > [2] There exists more than one negotiation protocol, but the > particular one from RFC 2543 is required to be supported by all SIP > user agents. > > [3] Every non-trivial software has bugs, and Ekiga isn't an exception. > Neither is Asterisk. In addition to that, developers have a tendency > to omit some obscure corner cases (it works for the most folks, but > you never know when yours will be the corner case not implemented), > and to interpret the same thing differently (leading to compatibility > issues). > _______________________________________________ > ekiga-list mailing list > [email protected] > http://mail.gnome.org/mailman/listinfo/ekiga-list
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