Hi, Just one question. The documentation I have seen says that the RTP audio stream is routed directly(if allowed ...), but never anything about video streams? Is this just because documents are pre 1.2 or is it true that audio can go directly, but video must pass through Asterisk? Anyone? Does anyone have experience with H263 on the 1.2.rc1 version? I think there is a bug, and will trace and submit it to Bugzilla..?? Trond
Trond Andersen wrote:> Just one question. The documentation I have seen says that the RTP > audio stream is routed directly(if allowed ...), but never anything > about video streams? Is this just because documents are pre 1.2 or is it > true that audio can go directly, but video must pass through Asterisk?All RTP streams are handled identically, regardless of their content.
Hi all iam working with * just started can some one explain me canreinvite=yes when should i use the above options I would like to use my * server for authentication and directly talk SIP user to SIP user with out consuming my * bandwidth, is that correct Does any one know, which provider support this option ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/0bad7c48/attachment.htm
canreinvite = yes tells the phones to try and talk to each other and leave Asterisk out of the mix. The important word here is TRY. There are lots of reasons that it might not quite work, and there was a big discussion on the list about it a little while ago. PaulH On Thu, 2006-03-02 at 01:55 +0530, ram wrote:> Hi all > > iam working with * just started > > can some one explain me canreinvite=yes > > when should i use the above options > > I would like to use my * server for authentication and directly talk > SIP user to SIP user > with out consuming my * bandwidth, is that correct > > Does any one know, which provider support this option > > ram > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi thanks, would mind pointing to me that let me check and see is that discussion will help me ram On 3/2/06, Paul Hales <pdhales@optusnet.com.au> wrote:> > > canreinvite = yes tells the phones to try and talk to each other and > leave Asterisk out of the mix. > > The important word here is TRY. > > There are lots of reasons that it might not quite work, and there was a > big discussion on the list about it a little while ago. > > PaulH > > On Thu, 2006-03-02 at 01:55 +0530, ram wrote: > > Hi all > > > > iam working with * just started > > > > can some one explain me canreinvite=yes > > > > when should i use the above options > > > > I would like to use my * server for authentication and directly talk > > SIP user to SIP user > > with out consuming my * bandwidth, is that correct > > > > Does any one know, which provider support this option > > > > ram > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/f7c2fedd/attachment.htm