[Asterisk-Dev] Peer to Peer Rtp stream

[dudlik]

and You shouldn't use cmd Dial with t or T parameters

dudl

Dne 6/1/2005, napsal "Brian West" <brian at bkw.org>:

>That shouldn't be the case if you set the global canreinvite=yes (which I
>think is the default)
>
>bkw
>
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
>> bounces at lists.digium.com] On Behalf Of Comer
>> Sent: Wednesday, January 05, 2005 9:37 PM
>> To: asterisk-dev at lists.digium.com
>> Subject: [Asterisk-Dev] Peer to Peer Rtp stream
>>
>> Hello,everyone
>> To two sip UAs who have registered on Asterisk,If  it is possible(for
>> example,no nat between them ),the RTP stream between them should be peer
>> to peer.
>> But As I know , If the two sip UAs accouts is store in Database
>> mysql(sipfriends),Not in sip.conf,the RTP stream will be relayed by
>> Asterisk.
>> Can anyone tell me why it is?
>> Thanks!
>>
>> $B!!!!!!!!!!!!!!!!(BComer
>> $B!!!!!!!!!!!!!!!!(Bhbyang at ustc.edu
>> $B!!!!!!!!!!!!!!!!!!!!(B2005-01-06
>>
>>