pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.

This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.

ASTERISK-25259 #close

Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
This commit is contained in:
Joshua Colp
2015-07-18 13:16:10 -03:00
parent a105461f9e
commit 309dd2a409
12 changed files with 147 additions and 8 deletions

View File

@@ -115,10 +115,6 @@ static int send_keepalive(const void *data)
time_t interval;
int send_keepalive;
if (!rtp) {
return 0;
}
keepalive = ast_rtp_instance_get_keepalive(rtp);
if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
@@ -140,6 +136,37 @@ static int send_keepalive(const void *data)
return (keepalive - interval) * 1000;
}
/*! \brief Check whether RTP is being received or not */
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;
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);
ast_softhangup(chan, AST_SOFTHANGUP_DEV);
ast_channel_unref(chan);
return 0;
}
/*! \brief Internal function which creates an RTP instance */
static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
{
@@ -174,6 +201,8 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
session->endpoint->media.cos_video, "SIP RTP Video");
}
ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
return 0;
}
@@ -1272,6 +1301,28 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
session_media, 1);
}
/* As the channel lock is not held during this process the scheduled item won't block if
* it is hanging up the channel at the same point we are applying this negotiated SDP.
*/
AST_SCHED_DEL(sched, session_media->timeout_sched_id);
/* Due to the fact that we only ever have one scheduled timeout item for when we are both
* off hold and on hold we don't need to store the two timeouts differently on the RTP
* instance itself.
*/
ast_rtp_instance_set_timeout(session_media->rtp, 0);
if (session->endpoint->media.rtp.timeout && !session_media->remotely_held) {
ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout);
} else if (session->endpoint->media.rtp.timeout_hold && session_media->remotely_held) {
ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout_hold);
}
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, 1);
}
return 1;
}
@@ -1301,9 +1352,8 @@ static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struc
static void stream_destroy(struct ast_sip_session_media *session_media)
{
if (session_media->rtp) {
if (session_media->keepalive_sched_id != -1) {
AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
}
AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
AST_SCHED_DEL(sched, session_media->timeout_sched_id);
ast_rtp_instance_stop(session_media->rtp);
ast_rtp_instance_destroy(session_media->rtp);
}