I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? Thank you very much.
John A. Sullivan III
2009-Nov-13 14:03 UTC
[asterisk-users] RTP traffic through Asterisk??
On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:> I have just established a call between 2 sip phones and I have noticed > that all RTP traffic goes through Asterisk Server. > > I was expecting RTP traffic went to one phone to another phone directly. > > I set canreinvite=yes in sip.conf in both sip peers. > > I also tested it with 2 mgcp phones and same result, all rtp traffic > goes through Asterisk. > > Is there any way to force traffic to go from one phone to another?<snip> I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society