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chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active

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Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).

ASTERISK-28774
Reported-by: Michael Neuhauser

Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
This commit is contained in:
Michael Neuhauser
2020-03-06 17:50:00 +01:00
committed by George Joseph
parent 82c3939c38
commit 5562fb2ea0
2 changed files with 41 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
channels/chan_pjsip.c
View File

@@ -332,6 +332,14 @@ static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instan
ast_sockaddr_setnull(&media->direct_media_addr);
changed = 1;
if (media->rtp) {
/* Direct media has ended - reset time of last received RTP packet
* to avoid premature RTP timeout. Synchronisation between the
* modification of direct_mdedia_addr+last_rx here and reading the
* values in res_pjsip_sdp_rtp.c:rtp_check_timeout() is provided
* by the channel's lock (which is held while this function is
* executed).
*/
ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
if (position != -1) {
ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1));

43
res/res_pjsip_sdp_rtp.c
View File

@@ -106,30 +106,53 @@ static int rtp_check_timeout(const void *data)
struct ast_sip_session_media *session_media = (struct ast_sip_session_media *)data;
struct ast_rtp_instance *rtp = session_media->rtp;
int elapsed;
int timeout;
struct ast_channel *chan;
if (!rtp) {
return 0;
}
elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
if (elapsed < ast_rtp_instance_get_timeout(rtp)) {
return (ast_rtp_instance_get_timeout(rtp) - elapsed) * 1000;
}
chan = ast_channel_get_by_name(ast_rtp_instance_get_channel_id(rtp));
if (!chan) {
return 0;
}
ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of RTP activity in %d seconds\n",
ast_channel_name(chan), elapsed);
/* Get channel lock to make sure that we access a consistent set of values
* (last_rx and direct_media_addr) - the lock is held when values are modified
* (see send_direct_media_request()/check_for_rtp_changes() in chan_pjsip.c). We
* are trying to avoid a situation where direct_media_addr has been reset but the
* last-rx time was not set yet.
*/
ast_channel_lock(chan);
ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
ast_channel_unlock(chan);
elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
timeout = ast_rtp_instance_get_timeout(rtp);
if (elapsed < timeout) {
ast_channel_unlock(chan);
ast_channel_unref(chan);
return (timeout - elapsed) * 1000;
}
/* Last RTP packet was received too long ago
* - disconnect channel unless direct media is in use.
*/
if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
ast_debug(3, "Not disconnecting channel '%s' for lack of %s RTP activity in %d seconds "
"since direct media is in use\n", ast_channel_name(chan),
ast_codec_media_type2str(session_media->type), elapsed);
ast_channel_unlock(chan);
ast_channel_unref(chan);
return timeout * 1000; /* recheck later, direct media may have ended then */
}
ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of %s RTP activity in %d seconds\n",
ast_channel_name(chan), ast_codec_media_type2str(session_media->type), elapsed);
ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
ast_softhangup(chan, AST_SOFTHANGUP_DEV);
ast_channel_unlock(chan);
ast_channel_unref(chan);
return 0;