diff options
| author | Richard Mudgett <rmudgett@digium.com> | 2015-03-24 19:26:11 +0000 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2015-03-24 19:26:11 +0000 |
| commit | b1e9552b087c68d412d6233034a1ad6e4eda02bd (patch) | |
| tree | ab61915028cc122d160f4b6e6161104c0b8017d7 /res/res_pjsip.c | |
| parent | a3fe43fbdc89aa51e266360dc93ed4a4445bebdb (diff) | |
chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip.c')
| -rw-r--r-- | res/res_pjsip.c | 23 |
1 files changed, 22 insertions, 1 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 1d57a072b..316a9472e 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -199,7 +199,7 @@ <para>This setting allows to choose the DTMF mode for endpoint communication.</para> <enumlist> <enum name="rfc4733"> - <para>DTMF is sent out of band of the main audio stream.This + <para>DTMF is sent out of band of the main audio stream. This supercedes the older <emphasis>RFC-2833</emphasis> used within the older <literal>chan_sip</literal>.</para> </enum> @@ -316,6 +316,27 @@ <configOption name="send_rpid" default="no"> <synopsis>Send the Remote-Party-ID header</synopsis> </configOption> + <configOption name="rpid_immediate" default="no"> + <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis> + <description> + <para>When enabled, immediately send <emphasis>180 Ringing</emphasis> + or <emphasis>183 Progress</emphasis> response messages to the + caller if the connected line information is updated before + the call is answered. This can send a <emphasis>180 Ringing</emphasis> + response before the call has even reached the far end. The + caller can start hearing ringback before the far end even gets + the call. Many phones tend to grab the first connected line + information and refuse to update the display if it changes. The + first information is not likely to be correct if the call + goes to an endpoint not under the control of this Asterisk + box.</para> + <para>When disabled, a connected line update must wait for + another reason to send a message with the connected line + information to the caller before the call is answered. You can + trigger the sending of the information by using an appropriate + dialplan application such as <emphasis>Ringing</emphasis>.</para> + </description> + </configOption> <configOption name="timers_min_se" default="90"> <synopsis>Minimum session timers expiration period</synopsis> <description><para> |