<div dir="ltr"><div dir="ltr">Hi, I wonder why the HTML documentation is not updated yet ?<div><br></div><div>when I test with the following command I do not detect any problem, is there any pipeline failing ?</div><div><font face="monospace, monospace"><br></font></div><div><font face="monospace, monospace">make modules-doc doc_format=html modules=modules/rtp_media_server</font><br></div><div><font face="monospace, monospace"><br></font></div><div><font face="monospace, monospace">Thanks !</font></div><div><font face="monospace, monospace"><br></font></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Feb 22, 2019 at 9:32 AM Kamailio Dev <<a href="mailto:kamailio.dev@kamailio.org">kamailio.dev@kamailio.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Module: kamailio<br>
Branch: master<br>
Commit: 4b7e6089e32ed71897396b95fed60b2461f14434<br>
URL: <a href="https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434" rel="noreferrer" target="_blank">https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434</a><br>
<br>
Author: Kamailio Dev <<a href="mailto:kamailio.dev@kamailio.org" target="_blank">kamailio.dev@kamailio.org</a>><br>
Committer: Kamailio Dev <<a href="mailto:kamailio.dev@kamailio.org" target="_blank">kamailio.dev@kamailio.org</a>><br>
Date: 2019-02-22T18:31:45+01:00<br>
<br>
modules: readme files regenerated - rtp_media_server ... [skip ci]<br>
<br>
---<br>
<br>
Modified: src/modules/rtp_media_server/README<br>
<br>
---<br>
<br>
Diff: <a href="https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.diff" rel="noreferrer" target="_blank">https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.diff</a><br>
Patch: <a href="https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.patch" rel="noreferrer" target="_blank">https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.patch</a><br>
<br>
---<br>
<br>
diff --git a/src/modules/rtp_media_server/README b/src/modules/rtp_media_server/README<br>
index bc47d7311e..742264f366 100644<br>
--- a/src/modules/rtp_media_server/README<br>
+++ b/src/modules/rtp_media_server/README<br>
@@ -1,4 +1,3 @@<br>
-<br>
rtp_media_server Module<br>
<br>
Julien Chavanton<br>
@@ -38,8 +37,9 @@ Julien Chavanton<br>
<br>
4.1. rms_answer ()<br>
4.2. rms_hangup ()<br>
- 4.3. rms_media_stop ()<br>
- 4.4. rms_play ()<br>
+ 4.3. rms_session_check ()<br>
+ 4.4. rms_sip_request ()<br>
+ 4.5. rms_play ()<br>
<br>
List of Examples<br>
<br>
@@ -48,6 +48,7 @@ Julien Chavanton<br>
1.3. usage example<br>
1.4. usage example<br>
1.5. usage example<br>
+ 1.6. usage example<br>
<br>
Chapter 1. Admin Guide<br>
<br>
@@ -67,8 +68,9 @@ Chapter 1. Admin Guide<br>
<br>
4.1. rms_answer ()<br>
4.2. rms_hangup ()<br>
- 4.3. rms_media_stop ()<br>
- 4.4. rms_play ()<br>
+ 4.3. rms_session_check ()<br>
+ 4.4. rms_sip_request ()<br>
+ 4.5. rms_play ()<br>
<br>
1. Overview<br>
<br>
@@ -111,6 +113,10 @@ Chapter 1. Admin Guide<br>
* mediastreamer2 git clone git://<a href="http://git.linphone.org/mediastreamer2.git" rel="noreferrer" target="_blank">git.linphone.org/mediastreamer2.git</a><br>
Mediastreamer2 is a powerful and lightweight streaming engine<br>
specialized for voice/video telephony applications.<br>
+ * bcunit git clone<br>
+ <a href="https://github.com/BelledonneCommunications/bcunit.git" rel="noreferrer" target="_blank">https://github.com/BelledonneCommunications/bcunit.git</a><br>
+ fork of the defunct project CUnit, with several fixes and patches<br>
+ applied. CUnit is a Unit testing framework for C.<br>
<br>
3. Parameters<br>
<br>
@@ -132,8 +138,9 @@ modparam("rtp_media_server", "log_file_name", "/var/log/rms/rms_ortp.log")<br>
<br>
4.1. rms_answer ()<br>
4.2. rms_hangup ()<br>
- 4.3. rms_media_stop ()<br>
- 4.4. rms_play ()<br>
+ 4.3. rms_session_check ()<br>
+ 4.4. rms_sip_request ()<br>
+ 4.5. rms_play ()<br>
<br>
4.1. rms_answer ()<br>
<br>
@@ -166,11 +173,7 @@ route {<br>
t_reply("503", "server error");<br>
}<br>
}<br>
-<br>
- if (is_method("BYE")){<br>
- xnotice("BYE RECEIVED [$ci]\n");<br>
- rms_media_stop();<br>
- }<br>
+ rms_sip_request();<br>
...<br>
<br>
4.2. rms_hangup ()<br>
@@ -184,10 +187,27 @@ route {<br>
rms_hangup();<br>
...<br>
<br>
-4.3. rms_media_stop ()<br>
+4.3. rms_session_check ()<br>
+<br>
+ Returns true if the current SIP message it handled/known by the RMS<br>
+ module, else it may be handle in any other way by Kamailio.<br>
+<br>
+ This function can be used from REQUEST_ROUTE, REPLY_ROUTE and<br>
+ FAILURE_ROUTE.<br>
+<br>
+ Example 1.4. usage example<br>
+...<br>
+ if (rms_session_check()) {<br>
+ xnotice("This session is handled by the RMS module\n");<br>
+ rms_sip_request();<br>
+ }<br>
+...<br>
+<br>
+4.4. rms_sip_request ()<br>
<br>
- This should be called on reception of a BYE, this will delete the RTP<br>
- session and the media ressources. and reply "200 OK".<br>
+ This should be called for every in-dialog SIP request, it will be<br>
+ forwarded behaving as a B2BUA, the transaction will be suspended until<br>
+ the second leg replies.<br>
<br>
If the SIP session is not found "481 Call/Transaction Does Not Exist"<br>
is returned.<br>
@@ -195,14 +215,14 @@ route {<br>
This function can be used from REQUEST_ROUTE, REPLY_ROUTE and<br>
FAILURE_ROUTE.<br>
<br>
- Example 1.4. usage example<br>
+ Example 1.5. usage example<br>
...<br>
- if (is_method("BYE")){<br>
- rms_media_stop();<br>
+ if (rms_session_check()) {<br>
+ rms_sip_request();<br>
}<br>
...<br>
<br>
-4.4. rms_play ()<br>
+4.5. rms_play ()<br>
<br>
Play a wav file, a resampler is automaticaly configured to resample and<br>
convert stereo to mono if needed.<br>
@@ -212,7 +232,7 @@ route {<br>
<br>
This function can be used from EVENT_ROUTE.<br>
<br>
- Example 1.5. usage example<br>
+ Example 1.6. usage example<br>
...<br>
rms_play("file.wav", "event_route_name");<br>
...<br>
<br>
<br>
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</blockquote></div>