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res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity

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Change RTP timer behavior for detecting RTP only after two-way
SDP channel establishment. Ignore detecting after receiving 183
with SDP or while direct media is used.
Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
and rtpholdtimeout options in chan_sip.

ASTERISK-26689 #close
ASTERISK-29929 #close

Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb
This commit is contained in:
Boris P. Korzun
2022-02-22 23:51:02 +03:00
committed by Joshua Colp
parent e5e02f783d
commit 82dbfe7783
2 changed files with 23 additions and 35 deletions
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50
res/res_pjsip_sdp_rtp.c
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@@ -106,9 +106,10 @@ 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;
int elapsed;
int now;
int timeout;
if (!rtp) {
return 0;
@@ -119,41 +120,37 @@ static int rtp_check_timeout(const void *data)
return 0;
}
/* 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);
elapsed = time(NULL) - ast_rtp_instance_get_last_rx(rtp);
/* Store these values locally to avoid multiple function calls */
now = time(NULL);
timeout = ast_rtp_instance_get_timeout(rtp);
if (elapsed < timeout) {
ast_channel_unlock(chan);
/* If the channel is not in UP state or call is redirected
* outside Asterisk return for later check.
*/
if (ast_channel_state(chan) != AST_STATE_UP || !ast_sockaddr_isnull(&session_media->direct_media_addr)) {
/* Avoiding immediately disconnect after channel up or direct media has been stopped */
ast_rtp_instance_set_last_rx(rtp, now);
ast_channel_unref(chan);
return (timeout - elapsed) * 1000;
/* Recheck after half timeout for avoiding possible races
* and faster reacting to cases while there is no an RTP at all.
*/
return timeout * 500;
}
/* 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_rtp(3, "(%p) RTP not disconnecting channel '%s' for lack of %s RTP activity in %d seconds "
"since direct media is in use\n", rtp, ast_channel_name(chan),
ast_codec_media_type2str(session_media->type), elapsed);
ast_channel_unlock(chan);
elapsed = now - ast_rtp_instance_get_last_rx(rtp);
if (elapsed < timeout) {
ast_channel_unref(chan);
return timeout * 1000; /* recheck later, direct media may have ended then */
return (timeout - elapsed) * 1000;
}
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_lock(chan);
ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
ast_softhangup(chan, AST_SOFTHANGUP_DEV);
ast_channel_unlock(chan);
ast_softhangup(chan, AST_SOFTHANGUP_DEV);
ast_channel_unref(chan);
return 0;
@@ -2234,8 +2231,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
}
if (ast_rtp_instance_get_timeout(session_media->rtp)) {
session_media->timeout_sched_id = ast_sched_add_variable(sched,
ast_rtp_instance_get_timeout(session_media->rtp) * 1000, rtp_check_timeout,
session_media->timeout_sched_id = ast_sched_add_variable(sched, 500, rtp_check_timeout,
session_media, 1);
}