In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan Karimi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041223/837bb7cc/attachment.htm
canreinvite=yes Aterisk stays in the signaling path so unless you're running tcpdump or the like you'll never notice this. bkw> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of bijan > Sent: Thursday, December 23, 2004 4:46 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] rtp channels not through asterisk > > In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > Currently with my settings, I notice that all rtp's are passing through my > asterisk. How could I achieve that they go directly from phone to phone? > I assume this way, my machine will have less load and therefore could > handle more calls. > > regards > Bijan Karimi >
> In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > > Currently with my settings, I notice that all rtps are passing through > my asterisk. How could I achieve that they go directly from phone to > phone? I assume this way, my machine will have less load and therefore > could handle more calls.As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work if the phones can reach each other directly (the phones and/or asterisk can't be behind a nat/firewall box).
Look at canreinvite= in the sip.conf. If you 'remove' Asterisk from the stream them you are using Asterisk more like a Proxy and less like a PBX. If this is the case and you want to support 'tons' of users look at something like SER. Asterisk is not a Sip proxy but rather a PBX and Media transcodeing gateway ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of bijan Sent: Thursday, December 23, 2004 5:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rtp channels not through asterisk In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan Karimi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041223/3d0e49cb/attachment.htm