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Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"

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This reverts commit 62745013a4.

Reason for revert: Issue when transcoding to/from g722

Change-Id: I1665a5442bfb6d7bfa06fdcea3374f4581395b4a
This commit is contained in:
George Joseph
2023-01-09 07:20:24 -06:00
parent bde2689e1b
commit 8067229418
8 changed files with 102 additions and 932 deletions
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19
channels/chan_pjsip.c
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@@ -2513,15 +2513,6 @@ 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
@@ -3002,16 +2993,6 @@ 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);