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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