res_hep_rtcp: Add module that sends RTCP information to a Homer Server

This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.

Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.

In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
   outbound calls. It now does this in the appropriate location, in the
   serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
   Specifically, some longs and unsigned ints can't be be packed into integer
   values, for obvious reasons. Since libjansson only supports integers,
   floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
   (a) it would emit a source IP address of 0.0.0.0 if bound to that IP
       address. We now use ast_find_ourip to get a better IP address, and
       properly marshal the result into an ast_strdupa'd string.
   (b) Reports can be generated with no report bodies. In particular, this
       occurs when a sender is transmitting information to a receiver (who
       will send no RTP back to the sender). As such, the sender has no report
       body for what it received. We now properly handle this case, and the
       sender will emit SR reports with no body. Likewise, if we receive an
       RTCP packet with no report body, we will still generate the appropriate
       events.

ASTERISK-24119 #close
........

Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 5 insertions, 0 deletions

diff --git a/CHANGES b/CHANGES
index a8564b845..299489e8c 100644
--- a/CHANGES
+++ b/CHANGES
@@ -245,6 +245,11 @@ res_pjsip
    created for an endpoint with this setting will have its accountcode set
    to the specified value.
 
+res_hep_rtcp
+------------------
+ * A new module, res_hep_rtcp, has been added that will forward RTCP call
+   statistics to a HEP capture server. See res_hep for more information.
+
 Functions
 ------------------
  * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now