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res_rtp_asterisk: Asterisk Media Experience Score (MES)

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This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

ASTERISK-30280

Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
This commit is contained in:
George Joseph
2022-10-28 04:57:56 -06:00
parent f0962d00ae
commit e66c5da145
8 changed files with 932 additions and 102 deletions
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19
channels/chan_pjsip.c
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@@ -2513,6 +2513,15 @@ static int hangup(void *data)
if (session) {
int cause = h_data->cause;
if (channel->session->active_media_state &&
channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
struct ast_sip_session_media *media =
channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
if (media->rtp) {
ast_rtp_instance_set_stats_vars(ast, media->rtp);
}
}
/*
* It's possible that session_terminate might cause the session to be destroyed
* immediately so we need to keep a reference to it so we can NULL session->channel
@@ -2993,6 +3002,16 @@ static void chan_pjsip_session_end(struct ast_sip_session *session)
SCOPE_EXIT_RTN("No channel\n");
}
if (session->active_media_state &&
session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
struct ast_sip_session_media *media =
session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
if (media->rtp) {
ast_rtp_instance_set_stats_vars(session->channel, media->rtp);
}
}
chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel));
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);