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-rw-r--r--channels/chan_h323.c6
-rw-r--r--channels/chan_mgcp.c5
-rw-r--r--channels/chan_sip.c34
-rw-r--r--channels/chan_skinny.c5
-rw-r--r--configs/sip.conf.sample4
-rw-r--r--include/asterisk/frame.h4
-rw-r--r--include/asterisk/rtp.h7
-rw-r--r--main/channel.c5
-rw-r--r--main/rtp.c57
9 files changed, 76 insertions, 51 deletions
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 700fdd174..f8a929d6e 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -918,7 +918,11 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(pvt->rtp);
+ ast_rtp_update_source(pvt->rtp);
+ res = 0;
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_change_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 159657b85..3f8b49e0d 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1442,7 +1442,10 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(sub->rtp);
+ ast_rtp_update_source(sub->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_change_source(sub->rtp);
break;
case -1:
transmit_notify_request(sub, "");
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 618fdd7ca..6d80d466f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -811,12 +811,11 @@ struct sip_auth {
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 28) /*!< 28: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 29) /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_RPORT_PRESENT (1 << 30) /*!< 30: Was rport received in the Via header? */
-#define SIP_PAGE2_CONSTANT_SSRC (1 << 31) /*!< 31: Don't change SSRC on reinvite */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
- SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_CONSTANT_SSRC)
+ SIP_PAGE2_UDPTL_DESTINATION)
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@@ -2952,9 +2951,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- ast_rtp_set_constantssrc(dialog->rtp);
- }
/* Set Frame packetization */
ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@@ -2965,9 +2961,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- ast_rtp_set_constantssrc(dialog->vrtp);
- }
}
ast_string_field_set(dialog, peername, peer->name);
@@ -3871,6 +3864,7 @@ static int sip_answer(struct ast_channel *ast)
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
+ ast_rtp_update_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -3905,7 +3899,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_rtp_new_source(p->rtp);
+ ast_rtp_update_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1);
@@ -4153,11 +4147,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_update_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_update_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -4168,7 +4162,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
res = -1;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(p->rtp);
+ ast_rtp_update_source(p->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_change_source(p->rtp);
break;
case -1:
res = -1;
@@ -15085,14 +15082,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
res = -1;
goto request_invite_cleanup;
}
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- if (p->rtp) {
- ast_rtp_set_constantssrc(p->rtp);
- }
- if (p->vrtp) {
- ast_rtp_set_constantssrc(p->vrtp);
- }
- }
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
if (option_debug > 1)
@@ -17574,9 +17563,6 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
- } else if (!strcasecmp(v->name, "constantssrc")) {
- ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
} else
res = 0;
@@ -18650,8 +18636,6 @@ static int reload_config(enum channelreloadreason reason)
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
} else if (!strcasecmp(v->name, "matchexterniplocally")) {
global_matchexterniplocally = ast_true(v->value);
- } else if (!strcasecmp(v->name, "constantssrc")) {
- ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
} else if (!strcasecmp(v->name, "shrinkcallerid")) {
if (ast_true(v->value)) {
global_shrinkcallerid = 1;
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 812aa16fb..48be77b9c 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -2868,7 +2868,10 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(sub->rtp);
+ ast_rtp_update_source(sub->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_change_source(sub->rtp);
break;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 914ccb775..838196d80 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -385,8 +385,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; instead of INVITE. This can be combined with 'nonat', as
; 'canreinvite=update,nonat'. It implies 'yes'.
-;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
-
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
@@ -546,7 +544,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; t38pt_usertpsource username
-; constantssrc template
; fromdomain
; regexten
; fromuser
@@ -559,7 +556,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; sendrpid
; outboundproxy
; rfc2833compensate
-; constantssrc
; t38pt_usertpsource
; contactpermit ; Limit what a host may register as (a neat trick
; contactdeny ; is to register at the same IP as a SIP provider,
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 5cfe0eb0a..7f45d91f0 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -85,7 +85,8 @@ struct ast_codec_pref {
\arg \b HOLD Call is placed on hold
\arg \b UNHOLD Call is back from hold
\arg \b VIDUPDATE Video update requested
- \arg \b SRCUPDATE The source of media has changed
+ \arg \b SRCUPDATE The source of media has changed (RTP marker bit must change)
+ \arg \b SRCCHANGE Media source has changed (RTP marker bit and SSRC must change)
*/
@@ -290,6 +291,7 @@ enum ast_control_frame_type {
AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */
AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */
AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */
+ AST_CONTROL_SRCCHANGE = 21, /*!< Media has changed and requires a new RTP SSRC */
};
#define AST_SMOOTHER_FLAG_G729 (1 << 0)
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index fada0bcb4..955e8bf99 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -179,10 +179,11 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
-/*! \brief When changing sources, don't generate a new SSRC */
-void ast_rtp_set_constantssrc(struct ast_rtp *rtp);
+/*! \brief Indicate that we need to set the marker bit */
+void ast_rtp_update_source(struct ast_rtp *rtp);
-void ast_rtp_new_source(struct ast_rtp *rtp);
+/*! \brief Indicate that we need to set the marker bit and change the ssrc */
+void ast_rtp_change_source(struct ast_rtp *rtp);
/*! \brief Setting RTP payload types from lines in a SDP description: */
void ast_rtp_pt_clear(struct ast_rtp* rtp);
diff --git a/main/channel.c b/main/channel.c
index a4ffd629d..4141d51bf 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -1972,6 +1972,7 @@ int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int cmdfd)
case AST_CONTROL_RINGING:
case AST_CONTROL_ANSWER:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
/* Unimportant */
break;
default:
@@ -2571,6 +2572,7 @@ static int attribute_const is_visible_indication(enum ast_control_frame_type con
case AST_CONTROL_PROCEEDING:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_RADIO_KEY:
case AST_CONTROL_RADIO_UNKEY:
case AST_CONTROL_OPTION:
@@ -2663,6 +2665,7 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
case AST_CONTROL_PROCEEDING:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_RADIO_KEY:
case AST_CONTROL_RADIO_UNKEY:
case AST_CONTROL_OPTION:
@@ -3367,6 +3370,7 @@ struct ast_channel *__ast_request_and_dial(const char *type, int format, void *d
case AST_CONTROL_UNHOLD:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case -1: /* Ignore -- just stopping indications */
break;
@@ -4316,6 +4320,7 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
case AST_CONTROL_UNHOLD:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
ast_indicate_data(other, f->subclass, f->data, f->datalen);
if (jb_in_use) {
ast_jb_empty_and_reset(c0, c1);
diff --git a/main/rtp.c b/main/rtp.c
index 4f54b71a5..76deb3c69 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -174,7 +174,6 @@ struct ast_rtp {
struct ast_codec_pref pref;
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
int set_marker_bit:1; /*!< Whether to set the marker bit or not */
- unsigned int constantssrc:1;
};
/* Forward declarations */
@@ -1175,6 +1174,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
unsigned int *rtpheader;
struct rtpPayloadType rtpPT;
struct ast_rtp *bridged = NULL;
+ AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
/* If time is up, kill it */
if (rtp->sending_digit)
@@ -1253,11 +1253,23 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
seqno &= 0xffff;
timestamp = ntohl(rtpheader[1]);
ssrc = ntohl(rtpheader[2]);
-
- if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
- if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
- mark = 1;
+
+ AST_LIST_HEAD_INIT_NOLOCK(&frames);
+ /* Force a marker bit and change SSRC if the SSRC changes */
+ if (rtp->rxssrc && rtp->rxssrc != ssrc) {
+ struct ast_frame *f, srcupdate = {
+ AST_FRAME_CONTROL,
+ .subclass = AST_CONTROL_SRCCHANGE,
+ };
+
+ if (!mark) {
+ if (option_debug || rtpdebug) {
+ ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
+ }
+ mark = 1;
+ }
+ f = ast_frisolate(&srcupdate);
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
}
rtp->rxssrc = ssrc;
@@ -1280,7 +1292,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
- return &ast_null_frame;
+ return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
}
rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */
@@ -1342,7 +1354,11 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
}
- return f ? f : &ast_null_frame;
+ if (f) {
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+ return AST_LIST_FIRST(&frames);
+ }
+ return &ast_null_frame;
}
rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
@@ -1358,7 +1374,8 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
- return f;
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+ return AST_LIST_FIRST(&frames);
}
}
@@ -1391,7 +1408,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
rtp->f.subclass |= 0x1;
}
rtp->f.src = "RTP";
- return &rtp->f;
+
+ AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
+ return AST_LIST_FIRST(&frames);
}
/* The following array defines the MIME Media type (and subtype) for each
@@ -2063,18 +2082,26 @@ int ast_rtp_settos(struct ast_rtp *rtp, int tos)
return res;
}
-void ast_rtp_set_constantssrc(struct ast_rtp *rtp)
+void ast_rtp_update_source(struct ast_rtp *rtp)
{
- rtp->constantssrc = 1;
+ if (rtp) {
+ rtp->set_marker_bit = 1;
+ if (option_debug > 2) {
+ ast_log(LOG_DEBUG, "Setting the marker bit due to a source update\n");
+ }
+ }
}
-void ast_rtp_new_source(struct ast_rtp *rtp)
+void ast_rtp_change_source(struct ast_rtp *rtp)
{
if (rtp) {
+ unsigned int ssrc = ast_random();
+
rtp->set_marker_bit = 1;
- if (!rtp->constantssrc) {
- rtp->ssrc = ast_random();
+ if (option_debug > 2) {
+ ast_log(LOG_DEBUG, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
}
+ rtp->ssrc = ssrc;
}
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-24 01:06:17 +0000