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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-07 10:45:24 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-09-07 10:45:24 +0000
commitb1d4a315309553c7a429598abcead42a6e428e37 (patch)
treef772c90945921f03f71bee542e9275c3043bb5c3 /configs/sip.conf.sample
parentf84257d3e57e02cdf3cebee89c1fd8415a201335 (diff)
Merged revisions 216438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216646 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample7
1 files changed, 7 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 7bfcda987..2b260eb25 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -208,6 +208,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
+;prematuremedia=no ; Some ISDN links send empty media frames before
+ ; the call is in ringing or progress state. The SIP
+ ; channel will then send 183 indicating early media
+ ; which will be empty - thus users get no ring signal.
+ ; Setting this to "no" will stop any media before we have
+ ; call progress. Default is "yes".
+
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it