You're right for pure SIP configurations , but Asterisk acts here as "media gateway" and treats all of the media comm. e.g. to eventually communicate with other types (like PSTN, H323, AIX, ... the voicemail app.) Michael Devenijn IT Manager DKMA Schaarbeeklei 636 B-1800 Vilvoorde Tel.: +32 2 255 10 19 Fax: +32 2 251 03 12 ________________________________ Van: Wim Venneman [mailto:wim.venneman@skynet.be] Verzonden: ma 8/12/2003 22:17 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] SIP (peer to peer?) Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP expert. Thank you for your help. Wim -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5169 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/0c369953/attachment.bin
Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP expert. Thank you for your help. Wim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/6fcc1b03/attachment.htm
SIP control messages goes always through the server (port 5060) , only RTP media streams is p2p . you can see RTP passing not p2p but by * server if: * the phone doesn't supports reinvites or * set in sip.conf canreinvite=no in the user definition Matteo. Il lun, 2003-12-08 alle 22:17, Wim Venneman ha scritto:> Hi all, > > Has anyone have an idea why, if you capture the files on a Asterisk > network (ex with Ethereal) you always see the communication between > the two sip phones( hard or soft) passing through the asterisk server > (on UDP layer) > > Isn't SIP a protocol that (after that it has established the call) , > he connects the two users with each other? > > > > Maybe a stupid question, but I'm not a SIP expert. > > > > Thank you for your help. > > > > Wim-- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl
Wim Venneman wrote:> > Has anyone have an idea why, if you capture the files on a Asterisk > network (ex with Ethereal) you always see the communication between the > two sip phones( hard or soft) passing through the asterisk server (on > UDP layer)Yes.> Isn't SIP a protocol that (after that it has established the call) , he > connects the two users with each other?Yes it is, but Asterisk is not really a SIP proxy, it's a PBX with SIP channels. Read here: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite /Olle
Hello! I think that's true. In older asterisk versions I saw such a "hand-over" between 2 sip phones and asterisk. But with the current versions I can't get it to work. I think you have to set "canreinvite=yes" at both clients that this can work. Additionally both ends need to have a common codec. If not asterisk has to stay between and convert codecs. But like I said I've problems with handover now. Has someone else encountered this loss of handovers in current asterisk versions? In my case I tried BudgetTone100 <-> Asterisk <->BudgetTone100. Wim Venneman wrote:> Hi all, > >Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) >Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? > > > >Maybe a stupid question, but I'm not a SIP expert. > > > >Thank you for your help. > > > >Wim > > > >
Depends if you're phone supports it, and you have reinvites etc enables in *. -d At 03:17 PM 12/8/2003, you wrote:> Hi all, > >Has anyone have an idea why, if you capture the files on a Asterisk >network (ex with Ethereal) you always see the communication between the >two sip phones( hard or soft) passing through the asterisk server (on UDP >layer)<?xml:namespace prefix = o ns = >"urn:schemas-microsoft-com:office:office" /> > >Isn't SIP a protocol that (after that it has established the call) , he >connects the two users with each other? > > > >Maybe a stupid question, but I'm not a SIP expert. > > > >Thank you for your help. > > > >Wim-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031210/32d01f4b/attachment.htm