[svn-commits] oej: trunk r48200 - in /trunk: ./ channels/ configs/
include/asterisk/ main/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Dec 2 05:05:41 MST 2006
Author: oej
Date: Sat Dec 2 06:05:40 2006
New Revision: 48200
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48200
Log:
- Disable RTP timeouts during T.38 transmission
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
trunk/include/asterisk/rtp.h
trunk/main/rtp.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=48200&r1=48199&r2=48200
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sat Dec 2 06:05:40 2006
@@ -961,8 +961,6 @@
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
- int rtpholdtimeout; /*!< RTP timeout when on hold */
- int rtpkeepalive; /*!< Send RTP packets for keepalive */
struct sockaddr_in recv; /*!< Received as */
struct in_addr ourip; /*!< Our IP */
struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
@@ -2726,17 +2724,21 @@
if (dialog->rtp) {
ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
+ /* Set Frame packetization */
+ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) {
ast_rtp_setdtmf(dialog->vrtp, 0);
ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
- }
-
- /* Set Frame packetization */
- if (dialog->rtp) {
- ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
- dialog->autoframing = peer->autoframing;
- }
+ ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
+ }
+
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
@@ -2774,8 +2776,6 @@
dialog->noncodeccapability &= ~AST_RTP_DTMF;
ast_string_field_set(dialog, context, peer->context);
dialog->rtptimeout = peer->rtptimeout;
- dialog->rtpholdtimeout = peer->rtpholdtimeout;
- dialog->rtpkeepalive = peer->rtpkeepalive;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
dialog->maxcallbitrate = peer->maxcallbitrate;
@@ -4299,16 +4299,19 @@
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_settos(p->rtp, global_tos_audio);
+ ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
if (p->vrtp) {
ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0);
ast_rtp_setdtmfcompensate(p->vrtp, 0);
+ ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
}
if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
p->maxcallbitrate = default_maxcallbitrate;
}
@@ -10427,6 +10430,7 @@
ast_cli(fd, " T1 minimum: %d\n", global_t1min);
ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
+ ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
@@ -11823,6 +11827,9 @@
if (bridgepvt->udptl) {
if (p->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
+ ast_rtp_set_rtptimers_onhold(p->rtp);
+ if (p->vrtp)
+ ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */
} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
@@ -14912,52 +14919,66 @@
/*! \brief helper function for the monitoring thread */
-static void check_rtp_timeout(struct sip_pvt *sip, time_t t)
-{
- if (sip->rtp && sip->owner &&
- (sip->owner->_state == AST_STATE_UP) &&
- !sip->redirip.sin_addr.s_addr) {
- if (sip->lastrtptx &&
- sip->rtpkeepalive &&
- (t > sip->lastrtptx + sip->rtpkeepalive)) {
- /* Need to send an empty RTP packet */
- sip->lastrtptx = time(NULL);
- ast_rtp_sendcng(sip->rtp, 0);
- }
- if (sip->lastrtprx &&
- (sip->rtptimeout || sip->rtpholdtimeout) &&
- (t > sip->lastrtprx + sip->rtptimeout)) {
- /* Might be a timeout now -- see if we're on hold */
- struct sockaddr_in sin;
- ast_rtp_get_peer(sip->rtp, &sin);
- if (sin.sin_addr.s_addr ||
- (sip->rtpholdtimeout &&
- (t > sip->lastrtprx + sip->rtpholdtimeout))) {
- /* Needs a hangup */
- if (sip->rtptimeout) {
- while (sip->owner && ast_channel_trylock(sip->owner)) {
- sip_pvt_unlock(sip);
- usleep(1);
- sip_pvt_lock(sip);
- }
- if (sip->owner) {
- if (!(ast_rtp_get_bridged(sip->rtp))) {
- ast_log(LOG_NOTICE,
- "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
- sip->owner->name,
- (long) (t - sip->lastrtprx));
- /* Issue a softhangup */
- ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
- } else
- ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", sip->owner->name, (long) (t - sip->lastrtprx));
- ast_channel_unlock(sip->owner);
- /* forget the timeouts for this call, since a hangup
- has already been requested and we don't want to
- repeatedly request hangups
- */
- sip->rtptimeout = 0;
- sip->rtpholdtimeout = 0;
- }
+static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
+{
+ /* If we have no RTP or no active owner, no need to check timers */
+ if (!dialog->rtp || !dialog->owner)
+ return;
+ /* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
+ if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
+ return;
+
+ /* If we have no timers set, return now */
+ if (ast_rtp_get_rtpkeepalive(dialog->rtp) == 0 || (ast_rtp_get_rtptimeout(dialog->rtp) == 0 && ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
+ return;
+
+ /* Check AUDIO RTP keepalives */
+ if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) &&
+ (t > dialog->lastrtptx + ast_rtp_get_rtpkeepalive(dialog->rtp))) {
+ /* Need to send an empty RTP packet */
+ dialog->lastrtptx = time(NULL);
+ ast_rtp_sendcng(dialog->rtp, 0);
+ }
+
+ /*! \todo Check video RTP keepalives
+
+ Do we need to move the lastrtptx to the RTP structure to have one for audio and one
+ for video? It really does belong to the RTP structure.
+ */
+
+ /* Check AUDIO RTP timers */
+ if (dialog->lastrtprx && (ast_rtp_get_rtptimeout(dialog->rtp) || ast_rtp_get_rtpholdtimeout(dialog->rtp)) &&
+ (t > dialog->lastrtprx + ast_rtp_get_rtptimeout(dialog->rtp))) {
+
+ /* Might be a timeout now -- see if we're on hold */
+ struct sockaddr_in sin;
+ ast_rtp_get_peer(dialog->rtp, &sin);
+ if (sin.sin_addr.s_addr || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
+ (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
+ /* Needs a hangup */
+ if (dialog->rtptimeout) {
+ while (dialog->owner && ast_channel_trylock(dialog->owner)) {
+ sip_pvt_unlock(dialog);
+ usleep(1);
+ sip_pvt_lock(dialog);
+ }
+ if (!(ast_rtp_get_bridged(dialog->rtp))) {
+ ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+ dialog->owner->name, (long) (t - dialog->lastrtprx));
+ /* Issue a softhangup */
+ ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
+ } else
+ ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", dialog->owner->name, (long) (t - dialog->lastrtprx));
+ ast_channel_unlock(dialog->owner);
+ /* forget the timeouts for this call, since a hangup
+ has already been requested and we don't want to
+ repeatedly request hangups
+ */
+ ast_rtp_set_rtptimeout(dialog->rtp, 0);
+ ast_rtp_set_rtpholdtimeout(dialog->rtp, 0);
+ if (dialog->vrtp) {
+ ast_rtp_set_rtptimeout(dialog->vrtp, 0);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, 0);
}
}
}
@@ -14971,7 +14992,7 @@
static void *do_monitor(void *data)
{
int res;
- struct sip_pvt *sip;
+ struct sip_pvt *dialog;
struct sip_peer *peer = NULL;
time_t t;
int fastrestart = FALSE;
@@ -14999,6 +15020,7 @@
if (sipsock > -1)
sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
}
+
/* Check for dialogs needing to be killed */
dialoglist_lock();
restartsearch:
@@ -15007,18 +15029,20 @@
of time since the last time we did it (when MWI is being sent, we can
get back to this point every millisecond or less)
*/
- for (sip = dialoglist; !fastrestart && sip; sip = sip->next) {
- sip_pvt_lock(sip);
+ for (dialog = dialoglist; !fastrestart && dialog; dialog = dialog->next) {
+ sip_pvt_lock(dialog);
/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
- check_rtp_timeout(sip, t);
+ check_rtp_timeout(dialog, t);
/* If we have sessions that needs to be destroyed, do it now */
- if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !sip->packets &&
- !sip->owner) {
- sip_pvt_unlock(sip);
- __sip_destroy(sip, TRUE, FALSE);
+ /* Check if we have outstanding requests not responsed to or an active call
+ - if that's the case, wait with destruction */
+ if (ast_test_flag(&dialog->flags[0], SIP_NEEDDESTROY) && !dialog->packets &&
+ !dialog->owner) {
+ sip_pvt_unlock(dialog);
+ __sip_destroy(dialog, TRUE, FALSE);
goto restartsearch;
}
- sip_pvt_unlock(sip);
+ sip_pvt_unlock(dialog);
}
dialoglist_unlock();
@@ -15027,7 +15051,8 @@
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
res = 1000;
- /* If we might need to send more mailboxes, don't wait long at all.*/
+
+ /* If we might need to send more mailbox notifications, don't wait long at all.*/
if (fastrestart)
res = 1;
res = ast_io_wait(io, res);
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=48200&r1=48199&r2=48200
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Sat Dec 2 06:05:40 2006
@@ -95,12 +95,6 @@
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
- ; when we're not on hold. This is to be able to hangup
- ; a call in the case of a phone disappearing from the net,
- ; like a powerloss or grandma tripping over a cable.
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
- ; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
@@ -162,6 +156,21 @@
;
;regcontext=sipregistrations
;
+;--------------------------- RTP timers ----------------------------------------------------
+; These timers are currently used for both audio and video streams. The RTP timeouts
+; are only applied to the audio channel.
+; The settings are settable in the global section as well as per device
+;
+;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're not on hold. This is to be able to hangup
+ ; a call in the case of a phone disappearing from the net,
+ ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're on hold (must be > rtptimeout)
+;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
+ ; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
@@ -206,8 +215,9 @@
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration (either general or
-; peer/user/friend sections)
+; both parties have T38 support enabled in their Asterisk configuration
+; This has to be enabled in the general section for all devices to work. You can then
+; disable it on a per device basis.
;
; t38pt_udptl = yes ; Default false
;
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?view=diff&rev=48200&r1=48199&r2=48200
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Sat Dec 2 06:05:40 2006
@@ -219,6 +219,21 @@
int ast_rtp_codec_getformat(int pt);
+/*! \brief Set rtp timeout */
+void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief Set rtp hold timeout */
+void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief set RTP keepalive interval */
+void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
+/*! \brief Get RTP keepalive interval */
+int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
+/*! \brief Get rtp hold timeout */
+int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
+/*! \brief Get rtp timeout */
+int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
+/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
+void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=48200&r1=48199&r2=48200
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Sat Dec 2 06:05:40 2006
@@ -128,6 +128,11 @@
double rxtransit; /*!< Relative transit time for previous packet */
int lasttxformat;
int lastrxformat;
+
+ int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+ int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+ int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
+
/* DTMF Reception Variables */
char resp;
unsigned int lasteventendseqn;
@@ -519,6 +524,53 @@
* Look in RFC 3550 Section A.7 for an example*/
interval = rtcpinterval;
return interval;
+}
+
+/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
+void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp)
+{
+ rtp->rtptimeout = (-1) * rtp->rtptimeout;
+ rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
+}
+
+/*! \brief Set rtp timeout */
+void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout)
+{
+ rtp->rtptimeout = timeout;
+}
+
+/*! \brief Set rtp hold timeout */
+void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout)
+{
+ rtp->rtpholdtimeout = timeout;
+}
+
+/*! \brief set RTP keepalive interval */
+void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period)
+{
+ rtp->rtpkeepalive = period;
+}
+
+/*! \brief Get rtp timeout */
+int ast_rtp_get_rtptimeout(struct ast_rtp *rtp)
+{
+ if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
+ return 0;
+ return rtp->rtptimeout;
+}
+
+/*! \brief Get rtp hold timeout */
+int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp)
+{
+ if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
+ return 0;
+ return rtp->rtpholdtimeout;
+}
+
+/*! \brief Get RTP keepalive interval */
+int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp)
+{
+ return rtp->rtpkeepalive;
}
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
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