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/1.6.0@252134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 697b90d306c1aab830a8b86334ddd994f1b62bf1..6408367315d701d0d4bbeee64a3cc488074a00c0 100644 (file)
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -919,7 +919,11 @@ 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 07cf590bb95e62508087f90b9e80df0641bf86a3..1e02db100b95974a234cf96d707296bb1ab1065d 100644 (file)
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1477,7 +1477,10 @@ 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 9e797e7d6f4945923275cad7bf02a2b839bd0518..e861f127b89b03b1ef46d8a33373c025eddd48a3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1008,13 +1008,12 @@ struct sip_auth {
 #define SIP_PAGE2_FAX_DETECT           (1 << 28)               /*!< DP: Fax Detection support */
 #define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
-#define SIP_PAGE2_CONSTANT_SSRC         (1 << 31)       /*!< GDP: Don't change SSRC on reinvite */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
        (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_UDPTL_DESTINATION | \
-        SIP_PAGE2_CONSTANT_SSRC | SIP_PAGE2_FAX_DETECT)
+        SIP_PAGE2_FAX_DETECT)
 
 /*@}*/ 
 
@@ -4333,9 +4332,6 @@ 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;
@@ -4346,9 +4342,6 @@ 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);
@@ -5322,7 +5315,7 @@ 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);
                ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
                res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
        }
@@ -5357,7 +5350,7 @@ 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);
@@ -5677,11 +5670,11 @@ 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 */
@@ -5700,7 +5693,10 @@ 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;
@@ -18168,14 +18164,6 @@ 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");
@@ -21367,9 +21355,6 @@ 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 if (!strcasecmp(v->name, "faxdetect")) {
                 ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
                 ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FAX_DETECT);
@@ -22878,8 +22863,6 @@ 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, "session-timers")) {
                        int i = (int) str2stmode(v->value); 
                        if (i < 0) {

diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index a04a564937a77da67e2d6789b4976344449f70b6..12533fbfc89ad582e408aa1e6e6625cdb6e6baa9 100644 (file)
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3769,7 +3769,10 @@ 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 141b29636675e6ae7adea419206072636f670f77..d832df6e3dc5ec94c0c96fc64c74d2cb1227aa0c 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -623,8 +623,6 @@ 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

diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index a5883cbaaf942eeafa1bc45200d678228d482afa..b054b49553bd8de698b4d3f1f525e9f18f508a10 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)
 
 */
 
@@ -302,6 +303,7 @@ 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 62af529c088d794acfbbb00e1bbadb8780c831a5..696fe7a9501edfb8624a667b10c0da59c890194d 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -187,10 +187,11 @@ 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 1ac9f3d5df47c71a9caf925c17d40a7c4fb6ee9f..c80ad1cf0579cf5e9d0a446156902b5f8b1de71f 100644 (file)
--- a/main/channel.c
+++ b/main/channel.c
@@ -2461,6 +2461,7 @@ 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:
@@ -3102,6 +3103,7 @@ 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:
@@ -3207,6 +3209,7 @@ 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:
@@ -3911,6 +3914,7 @@ 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;
 
@@ -4865,6 +4869,7 @@ 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, f->datalen);
                                if (jb_in_use) {

diff --git a/main/rtp.c b/main/rtp.c
index 393a9b1372dfa96a72ca69b0e7dcc4dca0ef05e2..5342a3e6e4ec30a017b0c8bf88961853e11244ff 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1434,6 +1434,7 @@ 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)
@@ -1533,10 +1534,22 @@ 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;
@@ -1567,7 +1580,7 @@ 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 */
@@ -1629,7 +1642,11 @@ 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;
@@ -1645,7 +1662,8 @@ 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);
                }
        }
 
@@ -1691,7 +1709,9 @@ 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
@@ -2382,18 +2402,22 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc)
        return ast_netsock_set_qos(rtp->s, tos, cos, 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;
        }
 }