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.dev@kamailio.org> wrote:
Module: kamailio
Branch: master
Commit: 4b7e6089e32ed71897396b95fed60b2461f14434
URL: https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434

Author: Kamailio Dev <kamailio.dev@kamailio.org>
Committer: Kamailio Dev <kamailio.dev@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");
 ...


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