diff options
author | Olle Johansson <oej@edvina.net> | 2006-12-02 12:05:40 +0000 |
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committer | Olle Johansson <oej@edvina.net> | 2006-12-02 12:05:40 +0000 |
commit | c23bc46089970c9c7275b662bbd48b0ed7310fc6 (patch) | |
tree | 09ffb0a36d6c9a88e86937e5c2cff44de6ff208f /configs | |
parent | eef9f7958bea41d443fb4e0698636459ab28446a (diff) |
- Disable RTP timeouts during T.38 transmission
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs')
-rw-r--r-- | configs/sip.conf.sample | 26 |
1 files changed, 18 insertions, 8 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index f706f2585..f4ba7e496 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -95,12 +95,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling -;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity - ; when we're not on hold. This is to be able to hangup - ; a call in the case of a phone disappearing from the net, - ; like a powerloss or grandma tripping over a cable. -;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity - ; when we're on hold (must be > rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;sendrpid = yes ; If Remote-Party-ID should be sent ;progressinband=never ; If we should generate in-band ringing always @@ -162,6 +156,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;regcontext=sipregistrations ; +;--------------------------- RTP timers ---------------------------------------------------- +; These timers are currently used for both audio and video streams. The RTP timeouts +; are only applied to the audio channel. +; The settings are settable in the global section as well as per device +; +;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity + ; on the audio channel + ; when we're not on hold. This is to be able to hangup + ; a call in the case of a phone disappearing from the net, + ; like a powerloss or grandma tripping over a cable. +;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity + ; on the audio channel + ; when we're on hold (must be > rtptimeout) +;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open + ; (default is off - zero) ;--------------------------- SIP DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration @@ -206,8 +215,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided -; both parties have T38 support enabled in their Asterisk configuration (either general or -; peer/user/friend sections) +; both parties have T38 support enabled in their Asterisk configuration +; This has to be enabled in the general section for all devices to work. You can then +; disable it on a per device basis. ; ; t38pt_udptl = yes ; Default false ; |