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

[Julien Chavanton]

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 at kamailio.org>
wrote:

> Module: kamailio
> Branch: master
> Commit: 4b7e6089e32ed71897396b95fed60b2461f14434
> URL:
> https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434
>
> Author: Kamailio Dev <kamailio.dev at kamailio.org>
> Committer: Kamailio Dev <kamailio.dev at 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 at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
>
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