[sr-dev] [kamailio/kamailio] rtp_media_server: adding call bridging (#1823)

[Julien Chavanton]

- forward in dialog messages using RMS bridge in B2BUA manner
- move all code related to rms_session_info to dedicated source file
- session_list refactoring to always include all dialog information
  needed
- suspend transaction to have non blocking bridging
- random rtp port
- stop using shared memory for MS2 and oRTP, manage all the allocation
  from one dedicated process

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#### Type Of Change
- [ ] Small bug fix (non-breaking change which fixes an issue)
- [ ] New feature (non-breaking change which adds new functionality)
- [ ] Breaking change (fix or feature that would change existing functionality)

#### Checklist:
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- [ ] PR should be backported to stable branches
- [x] Tested changes locally
- [ ] Related to issue #XXXX (replace XXXX with an open issue number)

#### Description
Even if this module is very much Alpha, I am getting closer to the minimal feature set I was having in mind, I should be able to focus my work on stabilizing the code and validate the behavior.

```
route {
        if (t_precheck_trans()) {
                t_check_trans();
                exit;
        }
        t_check_trans();
        xnotice("[$rm][$ci]\n");
        if (is_method("INVITE") && !has_totag()) {
                fix_nated_contact();
                if ($rU == "777") {
                        setflag(2);
                        if (rms_answer() != 1) {
                                send_reply("503", "server error");
                                xerr("rtp_media_server error!\n");
                                exit;
                        } else {
                                exit;
                        }
                }
                $var(target) = "sip:1234 at 11.12.69.51:5060;";
                if (!rms_bridge("$var(target)", "rms:bridged")) {
                        send_reply("604", "server error");
                        xerr("rtp_media_server bridge error!\n");
                        exit;
                }
        }
        if(rms_session_check())
                rms_sip_request();
        exit;
}
```


 
You can view, comment on, or merge this pull request online at:

  https://github.com/kamailio/kamailio/pull/1823

-- Commit Summary --

  * rtp_media_server: adding call bridging

-- File Changes --

    M src/core/dns_cache.c (1)
    M src/core/resolve.c (11)
    M src/modules/rtp_media_server/Makefile (0)
    M src/modules/rtp_media_server/config_example/kamailio.cfg (5)
    M src/modules/rtp_media_server/doc/rtp_media_server_admin.xml (40)
    A src/modules/rtp_media_server/doc/state.txt (14)
    M src/modules/rtp_media_server/rms_media.c (160)
    M src/modules/rtp_media_server/rms_media.h (7)
    M src/modules/rtp_media_server/rms_sdp.c (106)
    M src/modules/rtp_media_server/rms_sdp.h (2)
    A src/modules/rtp_media_server/rms_session_info.c (202)
    A src/modules/rtp_media_server/rms_session_info.h (109)
    M src/modules/rtp_media_server/rms_util.h (4)
    M src/modules/rtp_media_server/rtp_media_server.c (887)
    M src/modules/rtp_media_server/rtp_media_server.h (46)
    M src/modules/tm/ut.h (2)

-- Patch Links --

https://github.com/kamailio/kamailio/pull/1823.patch
https://github.com/kamailio/kamailio/pull/1823.diff

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