Merged revisions 252089 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines

  Only change the RTP ssrc when we see that it has changed

  This change basically reverts the change reviewed in
  https://reviewboard.asterisk.org/r/374/ and instead limits the
  updating of the RTP synchronization source to only those times when we
  detect that the other side of the conversation has changed the ssrc.

  The problem is that SRCUPDATE control frames are sent many times where
  we don't want a new ssrc, including whenever Asterisk has to send DTMF
  in a normal bridge. This is also not the first time that this mistake
  has been made. The initial implementation of the ast_rtp_new_source
  function also changed the ssrc--and then it was removed because of
  this same issue. Then, we put it back in again to fix a different
  issue. This patch attempts to only change the ssrc when we see that
  the other side of the conversation has changed the ssrc.

  It also renames some functions to make their purpose more clear.

  Review: https://reviewboard.asterisk.org/r/540/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@252137 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index be8c78d90a820faa5e8ae470e59e2dc0a07f08f5..5734fbb5846b66d3c7314ccce845e1574b20afad 100644 (file)
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -914,7 +914,11 @@ 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 771286da7be767c98da0327bed8fdce5832f4fe7..319ce910121067ba9cb126d281760f01aafe85ab 100644 (file)
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1454,7 +1454,10 @@ 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 6ab697768d5178eeb81c176850a2d6101571a32b..52dd81dac5240a51366b45c1bf6ad43bcc02c10a 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1373,7 +1373,6 @@ struct sip_auth {
 #define SIP_PAGE2_RTCACHEFRIENDS       (1 << 0)        /*!< GP: Should we keep RT objects in memory for extended time? */
 #define SIP_PAGE2_RTAUTOCLEAR          (1 << 2)        /*!< GP: Should we clean memory from peers after expiry? */
 /* Space for addition of other realtime flags in the future */
-#define SIP_PAGE2_CONSTANT_SSRC     (1 << 8)   /*!< GDP: Don't change SSRC on reinvite */
 #define SIP_PAGE2_STATECHANGEQUEUE     (1 << 9)        /*!< D: Unsent state pending change exists */
 
 #define SIP_PAGE2_RPORT_PRESENT         (1 << 10)       /*!< Was rport received in the Via header? */
@@ -1406,7 +1405,7 @@ struct sip_auth {
        (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
        SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
        SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
-       SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC)
+       SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS)
 
 /*@}*/ 
 
@@ -5150,9 +5149,6 @@ 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;
@@ -5163,9 +5159,6 @@ 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);
-               }
        }
        if (dialog->trtp) { /* Realtime text */
                ast_rtp_setdtmf(dialog->trtp, 0);
@@ -6195,7 +6188,7 @@ static int sip_answer(struct ast_channel *ast)
 
                ast_setstate(ast, AST_STATE_UP);
                ast_debug(1, "SIP answering channel: %s\n", ast->name);
-               ast_rtp_new_source(p->rtp);
+               ast_rtp_update_source(p->rtp);
                res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
                ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
        }
@@ -6230,7 +6223,7 @@ 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, TRUE);
@@ -6553,11 +6546,11 @@ 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 */
@@ -6576,7 +6569,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
                }
                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;
@@ -20002,14 +19998,6 @@ 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;
                        ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -23406,9 +23394,6 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
        } else if (!strcasecmp(v->name, "buggymwi")) {
                ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
                ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
-       } 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;
 
@@ -24877,8 +24862,6 @@ static int reload_config(enum channelreloadreason reason)
                                default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
                } else if (!strcasecmp(v->name, "matchexterniplocally")) {
                        sip_cfg.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, "session-timers")) {
                        int i = (int) str2stmode(v->value); 
                        if (i < 0) {

diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 4753e08e8c1a5524cdf16d82c306db5488a320db..c7341376a831d60937a605085f970d7d5474b6e9 100644 (file)
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -4256,7 +4256,10 @@ 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 e4e62ecb782383666d4c944fbd63237fd1a102dd..662ccc5f049eff4ca78f68a3c768f82d4d9952a8 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -696,8 +696,6 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; (observed with Microsoft OCS). By default this option is
                                 ; off.
 
-;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
@@ -910,7 +908,6 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; timerb
 ; qualifyfreq
 ; t38pt_usertpsource
-; constantssrc
 ; contactpermit         ; Limit what a host may register as (a neat trick
 ; contactdeny           ; is to register at the same IP as a SIP provider,
 ;                       ; then call oneself, and get redirected to that

diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 221548c97b966ce80b96a37bf599edde2e5c4d89..57760f162a478dfc91ccb1e29f2a3140eccc2693 100644 (file)
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -83,7 +83,8 @@ 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)
 
 */
 
@@ -316,6 +317,7 @@ enum ast_control_frame_type {
        _XXX_AST_CONTROL_T38 = 19,      /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
        AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
        AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */
+       AST_CONTROL_SRCCHANGE = 25,  /*!< Media source has changed and requires a new RTP SSRC */
 };
 
 enum ast_control_t38 {

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index da842e1f954498be7d301d3a9f5f6c1c791dd3c3..e7c80c58a5c9b3cb004b43eeb9ba538d79e6bad0 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -216,10 +216,11 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 
 int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
 
-/*! \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 cf2e7ff2816ea5f663bb78079d4d304f484b0390..f63063c626a86e22af2422885b2a6452c4ecb87a 100644 (file)
--- a/main/channel.c
+++ b/main/channel.c
@@ -2487,6 +2487,7 @@ 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:
@@ -3122,6 +3123,7 @@ 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:
@@ -3227,6 +3229,7 @@ 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:
@@ -3932,6 +3935,7 @@ 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;
 
@@ -4891,6 +4895,7 @@ 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:
                        case AST_CONTROL_T38_PARAMETERS:
                                ast_indicate_data(other, f->subclass, f->data.ptr, f->datalen);
                                if (jb_in_use) {

diff --git a/main/rtp.c b/main/rtp.c
index d463873f78ce689903c85f58f93689901a4cc647..c8e89d9a0ef6379e80a3260edf1a1033522d6275 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1578,6 +1578,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
        struct rtpPayloadType rtpPT;
        struct ast_rtp *bridged = NULL;
        int prev_seqno;
+       AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
        
        /* If time is up, kill it */
        if (rtp->sending_digit)
@@ -1679,10 +1680,22 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
        timestamp = ntohl(rtpheader[1]);
        ssrc = ntohl(rtpheader[2]);
        
-       if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
-               if (option_debug || rtpdebug)
-                       ast_debug(0, "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_debug(0, "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;
@@ -1713,7 +1726,7 @@ 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 */
@@ -1777,7 +1790,11 @@ 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_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
@@ -1793,7 +1810,8 @@ 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);
                }
        }
 
@@ -1897,7 +1915,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
                rtp->f.delivery.tv_usec = 0;
        }
        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
@@ -2638,18 +2658,22 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int type_of_service, int class_of_servic
        return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
 }
 
-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;
+               ast_debug(3, "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();
-               }
+               ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+               rtp->ssrc = ssrc;
        }
 }