Hello everyone, I?m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130220/81ead1b1/attachment.htm>
On Wed, 20 Feb 2013, Nguy?n C?ng wrote:> There is a phone call between two users, then they are talking to each > other directly or by the server. I mean all packets from the user A to > user B will be send directly to each other or will those packets from > user A must be send to server and server will send to user B.Depending on the technology (IAX or SIP) and the configuration, you can choose to have the Asterisk server handle the media (RTP) or not. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
2013/2/20 Nguy?n C?ng <nguyencong.1210 at gmail.com>> Hello everyone, I?m new to Asterisk and I have a question. There is a > phone call between two users, then they are talking to each other directly > or by the server. I mean all packets from the user A to user B will be send > directly to each other or will those packets from user A must be send to > server and server will send to user B.**** > > Thanks.**** > > -- >Both cases can happens. In a VoIP call we have two connections, one is used for signaling, usually port 5060 for SIP protocol, UDP transport and one is used for media (voice), usually random port. When the call starts the asterisk server sits in the middle of the media path, meaning all voice packets from phone A go to asterisk server and they are rerouted to phone B. After few milliseconds, if configured this way, asterisk server instructs the phone A to send the media directly to phone B to save bandwidth. It is named "reinvite" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130220/89d91223/attachment.htm>