Add support for using a jitterbuffer for RTP on bridged calls. This i…

…ncludes

a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)

Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@31052 f38db490-d61c-443f-a65b-d21fe96a405b

.cleancount

@@ -1 +1 @@
-17
+18

Makefile

@@ -282,7 +282,7 @@ OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
 	astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o \
 	utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
 	netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \
-	cryptostub.o sha1.o http.o
+	cryptostub.o sha1.o http.o scx_jitterbuf.o abstract_jb.o
 
 # we need to link in the objects statically, not as a library, because
 # otherwise modules will not have them available if none of the static

channel.c

@@ -1012,6 +1012,9 @@ void ast_channel_free(struct ast_channel *chan)
 	while ((vardata = AST_LIST_REMOVE_HEAD(headp, entries)))
 		ast_var_delete(vardata);
 
+	/* Destroy the jitterbuffer */
+	ast_jb_destroy(chan);
+
 	ast_string_field_free_all(chan);
 	free(chan);
 	AST_LIST_UNLOCK(&channels);
@@ -3303,6 +3306,9 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 	int watch_c0_dtmf;
 	int watch_c1_dtmf;
 	void *pvt0, *pvt1;
+	/* Indicates whether a frame was queued into a jitterbuffer */
+	int frame_put_in_jb = 0;
+	int jb_in_use;
 	int to;
 
 	cs[0] = c0;
@@ -3314,6 +3320,9 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 	watch_c0_dtmf = config->flags & AST_BRIDGE_DTMF_CHANNEL_0;
 	watch_c1_dtmf = config->flags & AST_BRIDGE_DTMF_CHANNEL_1;
 
+	/* Check the need of a jitterbuffer for each channel */
+	jb_in_use = ast_jb_do_usecheck(c0, c1);
+
 	for (;;) {
 		struct ast_channel *who, *other;
 
@@ -3332,9 +3341,15 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 			}
 		} else
 			to = -1;
+		/* Calculate the appropriate max sleep interval - in general, this is the time,
+		   left to the closest jb delivery moment */
+		if (jb_in_use)
+			to = ast_jb_get_when_to_wakeup(c0, c1, to);
 		who = ast_waitfor_n(cs, 2, &to);
 		if (!who) {
-			ast_log(LOG_DEBUG, "Nobody there, continuing...\n");
+			/* No frame received within the specified timeout - check if we have to deliver now */
+			if (jb_in_use)
+				ast_jb_get_and_deliver(c0, c1);
 			if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE || c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE) {
 				if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
 					c0->_softhangup = 0;
@@ -3354,6 +3369,9 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 		}
 
 		other = (who == c0) ? c1 : c0; /* the 'other' channel */
+		/* Try add the frame info the who's bridged channel jitterbuff */
+		if (jb_in_use)
+			frame_put_in_jb = !ast_jb_put(other, f);
 
 		if ((f->frametype == AST_FRAME_CONTROL) && !(config->flags & AST_BRIDGE_IGNORE_SIGS)) {
 			int bridge_exit = 0;
@@ -3390,8 +3408,13 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
 				ast_log(LOG_DEBUG, "Got DTMF on channel (%s)\n", who->name);
 				break;
 			}
-			/* other frames go to the other side */
-			ast_write(other, f);
+			/* Write immediately frames, not passed through jb */
+			if (!frame_put_in_jb)
+				ast_write(other, f);
+
+			/* Check if we have to deliver now */
+			if (jb_in_use)
+				ast_jb_get_and_deliver(c0, c1);
 		}
 		/* XXX do we want to pass on also frames not matched above ? */
 		ast_frfree(f);

channels/chan_alsa.c

@@ -70,6 +70,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "alsa-monitor.h"
 #endif
 
+#include "asterisk/abstract_jb.h"
+/* Global jitterbuffer configuration - by default, jb is disabled */
+static struct ast_jb_conf default_jbconf =
+{
+	.flags = 0,
+	.max_size = -1,
+	.resync_threshold = -1,
+	.impl = ""
+};
+static struct ast_jb_conf global_jbconf;
+
 #define DEBUG 0
 /* Which device to use */
 #define ALSA_INDEV "hw:0,0"
@@ -812,6 +823,8 @@ static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state)
 				tmp = NULL;
 			}
 		}
+		if (tmp)
+			ast_jb_configure(tmp, &global_jbconf);
 	}
 	return tmp;
 }
@@ -1051,9 +1064,18 @@ static int load_module(void *mod)
 	int x;
 	struct ast_config *cfg;
 	struct ast_variable *v;
+
+	/* Copy the default jb config over global_jbconf */
+	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+
 	if ((cfg = ast_config_load(config))) {
 		v = ast_variable_browse(cfg, "general");
 		while(v) {
+			/* handle jb conf */
+			if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
+				v = v->next;
+				continue;
+			}
 			if (!strcasecmp(v->name, "autoanswer"))
 				autoanswer = ast_true(v->value);
 			else if (!strcasecmp(v->name, "silencesuppression"))

channels/chan_iax2.c

@@ -1543,6 +1543,7 @@ static int __do_deliver(void *data)
 	  the IAX thread with the iaxsl lock held. */
 	struct iax_frame *fr = data;
 	fr->retrans = -1;
+	fr->af.has_timing_info = 0;
 	if (iaxs[fr->callno] && !ast_test_flag(iaxs[fr->callno], IAX_ALREADYGONE))
 		iax2_queue_frame(fr->callno, &fr->af);
 	/* Free our iax frame */

channels/chan_oss.c

@@ -82,6 +82,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "ring10.h"
 #include "answer.h"
 
+#include "asterisk/abstract_jb.h"
+/* Global jitterbuffer configuration - by default, jb is disabled */
+static struct ast_jb_conf default_jbconf =
+{
+	.flags = 0,
+	.max_size = -1,
+	.resync_threshold = -1,
+	.impl = ""
+};
+static struct ast_jb_conf global_jbconf;
+
 /*
  * Basic mode of operation:
  *
@@ -141,6 +152,33 @@ START_CONFIG
     ; queuesize = 10		; frames in device driver
     ; frags = 8			; argument to SETFRAGMENT
 
+    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+    ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
+                                  ; OSS channel. Defaults to "no". An enabled jitterbuffer will
+                                  ; be used only if the sending side can create and the receiving
+                                  ; side can not accept jitter. The ZAP channel can't accept jitter,
+                                  ; thus an enabled jitterbuffer on the receive ZAP side will always
+                                  ; be used if the sending side can create jitter or if ZAP jb is
+                                  ; forced.
+
+    ; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a ZAP
+                                  ; channel. Defaults to "no".
+
+    ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+    ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                                  ; resynchronized. Useful to improve the quality of the voice, with
+                                  ; big jumps in/broken timestamps, usualy sent from exotic devices
+                                  ; and programs. Defaults to 1000.
+
+    ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
+                                  ; channel. Two implementation are currenlty available - "fixed"
+                                  ; (with size always equals to jbmax-size) and "adaptive" (with
+                                  ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+    ;-----------------------------------------------------------------------------------
+
 [card1]
     ; device = /dev/dsp1	; alternate device
 
@@ -981,6 +1019,9 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *o,
 			/* XXX what about usecnt ? */
 		}
 	}
+	if (c)
+		ast_jb_configure(c, &global_jbconf);
+
 	return c;
 }
 
@@ -1407,6 +1448,10 @@ static struct chan_oss_pvt * store_config(struct ast_config *cfg, char *ctg)
 	for (v = ast_variable_browse(cfg, ctg);v; v=v->next) {
 		M_START(v->name, v->value);
 
+		/* handle jb conf */
+		if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
+			continue;
+
 		M_BOOL("autoanswer", o->autoanswer)
 		M_BOOL("autohangup", o->autohangup)
 		M_BOOL("overridecontext", o->overridecontext)
@@ -1472,6 +1517,9 @@ static int load_module(void *mod)
 	int i;
 	struct ast_config *cfg;
 
+	/* Copy the default jb config over global_jbconf */
+	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+
 	/* load config file */
 	cfg = ast_config_load(config);
 	if (cfg != NULL) {

channels/chan_sip.c

@@ -202,6 +202,17 @@ static int expiry = DEFAULT_EXPIRY;
 
 #define	INITIAL_CSEQ		101	/*!< our initial sip sequence number */
 
+#include "asterisk/abstract_jb.h"
+/* Global jitterbuffer configuration - by default, jb is disabled */
+static struct ast_jb_conf default_jbconf =
+{
+	.flags = 0,
+	.max_size = -1,
+	.resync_threshold = -1,
+	.impl = ""
+};
+static struct ast_jb_conf global_jbconf;
+
 static const char tdesc[] = "Session Initiation Protocol (SIP)";
 static const char config[] = "sip.conf";
 static const char notify_config[] = "sip_notify.conf";
@@ -850,6 +861,7 @@ static struct sip_pvt {
 	struct ast_variable *chanvars;		/*!< Channel variables to set for inbound call */
 	struct sip_pvt *next;			/*!< Next dialog in chain */
 	struct sip_invite_param *options;	/*!< Options for INVITE */
+	struct ast_jb_conf jbconf;
 } *iflist = NULL;
 
 #define FLAG_RESPONSE (1 << 0)
@@ -1256,7 +1268,7 @@ static const struct ast_channel_tech sip_tech = {
 	.type = "SIP",
 	.description = "Session Initiation Protocol (SIP)",
 	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
-	.properties = AST_CHAN_TP_WANTSJITTER,
+	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
 	.requester = sip_request_call,
 	.devicestate = sip_devicestate,
 	.call = sip_call,
@@ -3312,7 +3324,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 
 	if (recordhistory)
 		append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
-
+
+	/* Configure the new channel jb */
+	if (tmp && i && i->rtp)
+		ast_jb_configure(tmp, &i->jbconf);
+
 	return tmp;
 }
 
@@ -3647,6 +3663,9 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 		p->noncodeccapability |= AST_RTP_DTMF;
 	ast_string_field_set(p, context, default_context);
 
+	/* Assign default jb conf to the new sip_pvt */
+	memcpy(&p->jbconf, &global_jbconf, sizeof(struct ast_jb_conf));
+
 	/* Add to active dialog list */
 	ast_mutex_lock(&iflock);
 	p->next = iflist;
@@ -13247,7 +13266,6 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
 	for (; v; v = v->next) {
 		if (handle_common_options(&peerflags[0], &mask[0], v))
 			continue;
-
 		if (realtime && !strcasecmp(v->name, "regseconds")) {
 			ast_get_time_t(v->value, &regseconds, 0, NULL);
 		} else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
@@ -13540,12 +13558,19 @@ static int reload_config(enum channelreloadreason reason)
 	global_relaxdtmf = FALSE;
 	global_callevents = FALSE;
 	global_t1min = DEFAULT_T1MIN;		
+
+	/* Copy the default jb config over global_jbconf */
+	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT);
 
 	/* Read the [general] config section of sip.conf (or from realtime config) */
 	for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
 		if (handle_common_options(&global_flags[0], &dummy[0], v))
 			continue;
+		/* handle jb conf */
+		if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
+			continue;
 
 		/* Create the interface list */
 		if (!strcasecmp(v->name, "context")) {

channels/chan_zap.c

@@ -109,6 +109,17 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define SMDI_MD_WAIT_TIMEOUT 1500 /* 1.5 seconds */
 #endif
 
+#include "asterisk/abstract_jb.h"
+/* Global jitterbuffer configuration - by default, jb is disabled */
+static struct ast_jb_conf default_jbconf =
+{
+	.flags = 0,
+	.max_size = -1,
+	.resync_threshold = -1,
+	.impl = ""
+};
+static struct ast_jb_conf global_jbconf;
+
 #if !defined(ZT_SIG_EM_E1) || (defined(HAVE_LIBPRI) && !defined(ZT_SIG_HARDHDLC))
 #error "Your zaptel is too old.  please update"
 #endif
@@ -684,6 +695,7 @@ static struct zt_pvt {
 #endif	
 	int polarity;
 	int dsp_features;
+	struct ast_jb_conf jbconf;
 
 } *iflist = NULL, *ifend = NULL;
 
@@ -5195,6 +5207,9 @@ static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int
 		}
 	} else
 		ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
+	/* Configure the new channel jb */
+	if (tmp && i)
+		ast_jb_configure(tmp, &i->jbconf);
 	return tmp;
 }
 
@@ -6988,6 +7003,8 @@ static struct zt_pvt *mkintf(int channel, int signalling, int outsignalling, int
 		for (x=0;x<3;x++)
 			tmp->subs[x].zfd = -1;
 		tmp->channel = channel;
+		/* Assign default jb conf to the new zt_pvt */
+		memcpy(&tmp->jbconf, &global_jbconf, sizeof(struct ast_jb_conf));
 	}
 
 	if (tmp) {
@@ -10200,6 +10217,7 @@ static int setup_zap(int reload)
 {
 	struct ast_config *cfg;
 	struct ast_variable *v;
+	struct ast_variable *vjb;
 	struct zt_pvt *tmp;
 	char *chan;
 	char *c;
@@ -10289,6 +10307,11 @@ static int setup_zap(int reload)
 	}
 #endif
 	v = ast_variable_browse(cfg, "channels");
+	/* Copy the default jb config over global_jbconf */
+	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+	/* Traverse all variables to handle jb conf */
+	for (vjb = v; vjb; vjb = vjb->next)
+		ast_jb_read_conf(&global_jbconf, vjb->name, vjb->value);
 	while(v) {
 		/* Create the interface list */
 		if (!strcasecmp(v->name, "channel")