Author: rmudgett Date: Tue Mar 24 14:41:36 2015 New Revision: 433339 URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=433339 Log: 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/ ........ Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13 Modified: trunk/ (props changed) trunk/CHANGES trunk/channels/chan_pjsip.c trunk/configs/samples/pjsip.conf.sample trunk/include/asterisk/res_pjsip.h trunk/res/res_pjsip.c trunk/res/res_pjsip/pjsip_configuration.c Propchange: trunk/ ------------------------------------------------------------------------------ Binary property 'branch-13-merged' - no diff available. Modified: trunk/CHANGES URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/CHANGES (original) +++ trunk/CHANGES Tue Mar 24 14:41:36 2015 @@ -105,6 +105,18 @@ * Added preferchannelclass=no option to prefer the application-passed class over the channel-set musicclass. This allows separate hold-music from application (e.g. Queue or Dial) specified music. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------ +------------------------------------------------------------------------------ + +chan_pjsip +------------------ + * New 'rpid_immediate' option to control if connected line update information + goes to the caller immediately or waits for another reason to send the + connected line information update. See the online option documentation for + more information. Defaults to 'no' as setting it to 'yes' can result in + many unnecessary messages being sent to the caller. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------ Modified: trunk/channels/chan_pjsip.c URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_pjsip.c?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/channels/chan_pjsip.c (original) +++ trunk/channels/chan_pjsip.c Tue Mar 24 14:41:36 2015 @@ -1117,7 +1117,8 @@ ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp); } - } else if (session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED + } else if (session->endpoint->id.rpid_immediate + && session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED && is_colp_update_allowed(session)) { int response_code = 0; Modified: trunk/configs/samples/pjsip.conf.sample URL: http://svnview.digium.com/svn/asterisk/trunk/configs/samples/pjsip.conf.sample?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/configs/samples/pjsip.conf.sample (original) +++ trunk/configs/samples/pjsip.conf.sample Tue Mar 24 14:41:36 2015 @@ -637,6 +637,7 @@ ; information to the called user agent (default: "yes") ;send_pai=no ; Send the P Asserted Identity header (default: "no") ;send_rpid=no ; Send the Remote Party ID header (default: "no") +;rpid_immediate=no ; Send connected line updates on unanswered incoming calls immediately. (default: "no") ;timers_min_se=90 ; Minimum session timers expiration period (default: ; "90") ;timers=yes ; Session timers for SIP packets (default: "yes") Modified: trunk/include/asterisk/res_pjsip.h URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_pjsip.h?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/include/asterisk/res_pjsip.h (original) +++ trunk/include/asterisk/res_pjsip.h Tue Mar 24 14:41:36 2015 @@ -415,6 +415,8 @@ unsigned int send_pai; /*! Do we send Remote-Party-ID headers to this endpoint? */ unsigned int send_rpid; + /*! Do we send messages for connected line updates for unanswered incoming calls immediately to this endpoint? */ + unsigned int rpid_immediate; /*! Do we add Diversion headers to applicable outgoing requests/responses? */ unsigned int send_diversion; /*! When performing connected line update, which method should be used */ Modified: trunk/res/res_pjsip.c URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip.c?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/res/res_pjsip.c (original) +++ trunk/res/res_pjsip.c Tue Mar 24 14:41:36 2015 @@ -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> @@ -315,6 +315,27 @@ </configOption> <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> Modified: trunk/res/res_pjsip/pjsip_configuration.c URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip/pjsip_configuration.c?view=diff&rev=433339&r1=433338&r2=433339 ============================================================================== --- trunk/res/res_pjsip/pjsip_configuration.c (original) +++ trunk/res/res_pjsip/pjsip_configuration.c Tue Mar 24 14:41:36 2015 @@ -1710,6 +1710,7 @@ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "trust_id_outbound", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.trust_outbound)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_pai", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_pai)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_rpid", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_rpid)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rpid_immediate", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.rpid_immediate)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_diversion", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_diversion)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "mailboxes", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, subscription.mwi.mailboxes)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "aggregate_mwi", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, subscription.mwi.aggregate)); -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits
