diff options
Diffstat (limited to 'configs/sip.conf.sample')
| -rw-r--r-- | configs/sip.conf.sample | 32 |
1 files changed, 16 insertions, 16 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 7a43598d4..b2050879f 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -611,17 +611,17 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; (work around more UNIDEN bugs) ;----------------------------------- MEDIA HANDLING -------------------------------- -; By default, Asterisk tries to re-invite the audio to an optimal path. If there's +; By default, Asterisk tries to re-invite media streams to an optimal path. If there's ; no reason for Asterisk to stay in the media path, the media will be redirected. -; This does not really work with in the case where Asterisk is outside and have -; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat +; This does not really work well in the case where Asterisk is outside and the +; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. ; -;canreinvite=yes ; Asterisk by default tries to redirect the - ; RTP media stream (audio) to go directly from +;directmedia=yes ; Asterisk by default tries to redirect the + ; RTP media stream to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients - ; behind a NAT, or for some other reason wants Asterisk to + ; behind a NAT, or for some other reason want Asterisk to ; stay in the audio path, you may want to turn this off. ; This setting also affect direct RTP @@ -633,18 +633,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the - ; callers INVITE. This will also fail if canreinvite is enabled when + ; callers INVITE. This will also fail if directmedia is enabled when ; the device is actually behind NAT. -;canreinvite=nonat ; An additional option is to allow media path redirection +;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). -;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, +;directmedia=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. This can be combined with 'nonat', as - ; 'canreinvite=update,nonat'. It implies 'yes'. + ; 'directmedia=update,nonat'. It implies 'yes'. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version ; number in SDP packets and will only modify the SDP @@ -808,7 +808,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; remotesecret ; transport ; dtmfmode -; canreinvite +; directmedia ; nat ; callgroup ; pickupgroup @@ -918,12 +918,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes - canreinvite=no + directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options nat=no - canreinvite=yes + directmedia=yes [my-codecs](!) ; a template for my preferred codecs disallow=all @@ -958,7 +958,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;host=192.168.0.23 ; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk -;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk +;directmedia=yes ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk (deprecated) @@ -988,7 +988,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;callerid="Jane Smith" <5678> ;host=dynamic ; This device needs to register ;nat=yes ; X-Lite is behind a NAT router -;canreinvite=no ; Typically set to NO if behind NAT +;directmedia=no ; Typically set to NO if behind NAT ;disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw @@ -1061,7 +1061,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Send SIP and RTP to the IP address that packet is ; received from instead of trusting SIP headers ;host=dynamic ; This device registers with us -;canreinvite=no ; Asterisk by default tries to redirect the +;directmedia=no ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is |