[sr-dev] git:master:4b7e6089: modules: readme files regenerated - rtp_media_server ... [skip ci]

Module: kamailio Branch: master Commit: 4b7e6089e32ed71897396b95fed60b2461f14434 URL: https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2… Author: Kamailio Dev <kamailio.dev(a)kamailio.org> Committer: Kamailio Dev <kamailio.dev(a)kamailio.org> Date: 2019-02-22T18:31:45+01:00 modules: readme files regenerated - rtp_media_server ... [skip ci] --- Modified: src/modules/rtp_media_server/README --- Diff: https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2… Patch: https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2… --- diff --git a/src/modules/rtp_media_server/README b/src/modules/rtp_media_server/README index bc47d7311e..742264f366 100644 --- a/src/modules/rtp_media_server/README +++ b/src/modules/rtp_media_server/README @@ -1,4 +1,3 @@ - rtp_media_server Module Julien Chavanton @@ -38,8 +37,9 @@ Julien Chavanton 4.1. rms_answer () 4.2. rms_hangup () - 4.3. rms_media_stop () - 4.4. rms_play () + 4.3. rms_session_check () + 4.4. rms_sip_request () + 4.5. rms_play () List of Examples @@ -48,6 +48,7 @@ Julien Chavanton 1.3. usage example 1.4. usage example 1.5. usage example + 1.6. usage example Chapter 1. Admin Guide @@ -67,8 +68,9 @@ Chapter 1. Admin Guide 4.1. rms_answer () 4.2. rms_hangup () - 4.3. rms_media_stop () - 4.4. rms_play () + 4.3. rms_session_check () + 4.4. rms_sip_request () + 4.5. rms_play () 1. Overview @@ -111,6 +113,10 @@ Chapter 1. Admin Guide * mediastreamer2 git clone git://git.linphone.org/mediastreamer2.git Mediastreamer2 is a powerful and lightweight streaming engine specialized for voice/video telephony applications. + * bcunit git clone + https://github.com/BelledonneCommunications/bcunit.git + fork of the defunct project CUnit, with several fixes and patches + applied. CUnit is a Unit testing framework for C. 3. Parameters @@ -132,8 +138,9 @@ modparam("rtp_media_server", "log_file_name", "/var/log/rms/rms_ortp.log") 4.1. rms_answer () 4.2. rms_hangup () - 4.3. rms_media_stop () - 4.4. rms_play () + 4.3. rms_session_check () + 4.4. rms_sip_request () + 4.5. rms_play () 4.1. rms_answer () @@ -166,11 +173,7 @@ route { t_reply("503", "server error"); } } - - if (is_method("BYE")){ - xnotice("BYE RECEIVED [$ci]\n"); - rms_media_stop(); - } + rms_sip_request(); ... 4.2. rms_hangup () @@ -184,10 +187,27 @@ route { rms_hangup(); ... -4.3. rms_media_stop () +4.3. rms_session_check () + + Returns true if the current SIP message it handled/known by the RMS + module, else it may be handle in any other way by Kamailio. + + This function can be used from REQUEST_ROUTE, REPLY_ROUTE and + FAILURE_ROUTE. + + Example 1.4. usage example +... + if (rms_session_check()) { + xnotice("This session is handled by the RMS module\n"); + rms_sip_request(); + } +... + +4.4. rms_sip_request () - This should be called on reception of a BYE, this will delete the RTP - session and the media ressources. and reply "200 OK". + This should be called for every in-dialog SIP request, it will be + forwarded behaving as a B2BUA, the transaction will be suspended until + the second leg replies. If the SIP session is not found "481 Call/Transaction Does Not Exist" is returned. @@ -195,14 +215,14 @@ route { This function can be used from REQUEST_ROUTE, REPLY_ROUTE and FAILURE_ROUTE. - Example 1.4. usage example + Example 1.5. usage example ... - if (is_method("BYE")){ - rms_media_stop(); + if (rms_session_check()) { + rms_sip_request(); } ... -4.4. rms_play () +4.5. rms_play () Play a wav file, a resampler is automaticaly configured to resample and convert stereo to mono if needed. @@ -212,7 +232,7 @@ route { This function can be used from EVENT_ROUTE. - Example 1.5. usage example + Example 1.6. usage example ... rms_play("file.wav", "event_route_name"); ...