- Disable RTP hold timers while T.38 fax transmission happens

- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 18 insertions, 8 deletions

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 37e3ebba3..33959eff4 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -91,12 +91,6 @@ 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
@@ -152,6 +146,21 @@ 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
@@ -196,8 +205,9 @@ 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
 ;