Hi, I wonder why the HTML documentation is not updated yet ? when I test with the following command I do not detect any problem, is there any pipeline failing ?
make modules-doc doc_format=html modules=modules/rtp_media_server Thanks ! On Fri, Feb 22, 2019 at 9:32 AM Kamailio Dev <kamailio....@kamailio.org> wrote: > Module: kamailio > Branch: master > Commit: 4b7e6089e32ed71897396b95fed60b2461f14434 > URL: > https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434 > > Author: Kamailio Dev <kamailio....@kamailio.org> > Committer: Kamailio Dev <kamailio....@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/4b7e6089e32ed71897396b95fed60b2461f14434.diff > Patch: > https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.patch > > --- > > 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"); > ... > > > _______________________________________________ > Kamailio (SER) - Development Mailing List > sr-dev@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >
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