Merged revisions 207424 via svnmerge from

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

................
  r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines
  
  Merged revisions 207423 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines
    
    Answer video SDP offers properly when videosupport is not enabled.
    
    Copied from Review board:
    
    In issue 12434, the reporter describes a situation in which audio and video 
    is offered on the call, but because videosupport is disabled in sip.conf, 
    Asterisk gives no response at all to the video offer. According to RFC 3264, 
    all media offers should have a corresponding answer. For offers we do not 
    intend to actually reply to with meaningful values, we should still reply 
    with the port for the media stream set to 0.
    
    In this patch, we take note of what types of media have been offered and 
    save the information on the sip_pvt. The SDP in the response will take into 
    account whether media was offered. If we are not otherwise going to answer 
    a media offer, we will insert an appropriate m= line with the port set to 0.
    
    It is important to note that this patch is pretty much a bandage being 
    applied to a broken bone. The patch *only* helps for situations where video 
    is offered but videosupport is disabled and when udptl_pt is disabled but 
    T.38 is offered. Asterisk is not guaranteed to respond to every media offer. 
    Notable cases are when multiple streams of the same type are offered. 
    The 2 media stream limit is still present with this patch, too.
    
    In trunk and the 1.6.X branches, things will be a bit different since Asterisk 
    also supports text in SDPs as well.
    
    (closes issue #12434)
    Reported by: mnnojd
    
    Review: https://reviewboard.asterisk.org/r/311
    Review: https://reviewboard.asterisk.org/r/313
  ........
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@207426 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 51 insertions, 6 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 11b7c5244..777b5db9c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1246,8 +1246,10 @@ struct sip_st_cfg {
 	int st_max_se;                  /*!< Highest threshold for session refresh interval */
 };
 
-
-
+struct offered_media {
+	int offered;
+	char text[128];
+};
 
 /*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
  * Created and initialized by sip_alloc(), the descriptor goes into the list of
@@ -1404,6 +1406,21 @@ struct sip_pvt {
 	struct sip_st_dlg *stimer;		/*!< SIP Session-Timers */              
 	int red; 				/*!< T.140 RTP Redundancy */
 	int hangupcause;			/*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
+	/*! When receiving an SDP offer, it is important to take note of what media types were offered.
+	 * By doing this, even if we don't want to answer a particular media stream with something meaningful, we can
+	 * still put an m= line in our answer with the port set to 0.
+	 *
+	 * The reason for the length being 4 is that in this branch of Asterisk, the only media types supported are 
+	 * image, audio, text, and video. Therefore we need to keep track of which types of media were offered.
+	 *
+	 * Note that if we wanted to be 100% correct, we would keep a list of all media streams offered. That way we could respond
+	 * even to unknown media types, and we could respond to multiple streams of the same type. Such large-scale changes
+	 * are not a good idea for released branches, though, so we're compromising by just making sure that for the common cases:
+	 * audio and video, audio and T.38, and audio and text, we give the appropriate response to both media streams.
+	 *
+	 * The large-scale changes would be a good idea for implementing during an SDP rewrite.
+	 */
+	struct offered_media offered_media[4];
 }; 
 
 /*! Max entires in the history list for a sip_pvt */
@@ -7098,6 +7115,8 @@ static int find_sdp(struct sip_request *req)
 enum media_type {
 	SDP_AUDIO,
 	SDP_VIDEO,
+	SDP_IMAGE,
+	SDP_TEXT,
 };
 
 static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_type media, struct sockaddr_in *sin)
@@ -7251,6 +7270,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	/* Update our last rtprx when we receive an SDP, too */
 	p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
 
+	memset(p->offered_media, 0, sizeof(p->offered_media));
+
 	/* Store the SDP version number of remote UA. This will allow us to 
 	distinguish between session modifications and session refreshes. If 
 	the remote UA does not send an incremented SDP version number in a 
@@ -7369,11 +7390,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
 		    (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
 			audio = TRUE;
+			p->offered_media[SDP_AUDIO].offered = TRUE;
 			numberofmediastreams++;
 			/* Found audio stream in this media definition */
 			portno = x;
 			/* Scan through the RTP payload types specified in a "m=" line: */
-			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+			codecs = m + len;
+			ast_copy_string(p->offered_media[SDP_AUDIO].text, codecs, sizeof(p->offered_media[SDP_AUDIO].text));
+			for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
 				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
 					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
 					return -1;
@@ -7386,10 +7410,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		    (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
 			video = TRUE;
 			p->novideo = FALSE;
+			p->offered_media[SDP_VIDEO].offered = TRUE;
 			numberofmediastreams++;
 			vportno = x;
 			/* Scan through the RTP payload types specified in a "m=" line: */
-			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+			codecs = m + len;
+			ast_copy_string(p->offered_media[SDP_VIDEO].text, codecs, sizeof(p->offered_media[SDP_VIDEO].text));
+			for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
 				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
 					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
 					return -1;
@@ -7401,11 +7428,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		} else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
 		    (sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
 			text = TRUE;
+			p->offered_media[SDP_TEXT].offered = TRUE;
 			p->notext = FALSE;
 			numberofmediastreams++;
 			tportno = x;
 			/* Scan through the RTP payload types specified in a "m=" line: */
-			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+			codecs = m + len;
+			ast_copy_string(p->offered_media[SDP_TEXT].text, codecs, sizeof(p->offered_media[SDP_TEXT].text));
+			for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
 				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
 					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
 					return -1;
@@ -7418,6 +7448,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			(sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len > 0) )) {
 			if (debug)
 				ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
+			p->offered_media[SDP_IMAGE].offered = TRUE;
 			udptlportno = x;
 			numberofmediastreams++;
 		} else 
@@ -8294,7 +8325,6 @@ static inline int resp_needs_contact(const char *msg, enum sipmethod method) {
 	return 0;
 }
 
-
 /*! \brief Prepare SIP response packet */
 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
 {
@@ -8982,6 +9012,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 
 	char codecbuf[SIPBUFSIZE];
 	char buf[SIPBUFSIZE];
+	char dummy_answer[256];
 
 	/* Set the SDP session name */
 	snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
@@ -9256,20 +9287,31 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		add_line(resp, m_audio->str);
 		add_line(resp, a_audio->str);
 		add_line(resp, hold);
+	} else if (p->offered_media[SDP_AUDIO].offered) {
+		snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].text);
+		add_line(resp, dummy_answer);
 	}
 	if (needvideo) { /* only if video response is appropriate */
 		add_line(resp, m_video->str);
 		add_line(resp, a_video->str);
 		add_line(resp, hold);	/* Repeat hold for the video stream */
+	} else if (p->offered_media[SDP_VIDEO].offered) {
+		snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].text);
+		add_line(resp, dummy_answer);
 	}
 	if (needtext) { /* only if text response is appropriate */
 		add_line(resp, m_text->str);
 		add_line(resp, a_text->str);
 		add_line(resp, hold);	/* Repeat hold for the text stream */
+	} else if (p->offered_media[SDP_TEXT].offered) {
+		snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].text);
+		add_line(resp, dummy_answer);
 	}
 	if (add_t38) {
 		add_line(resp, m_modem->str);
 		add_line(resp, a_modem->str);
+	} else if (p->offered_media[SDP_IMAGE].offered) {
+		add_line(resp, "m=image 0 udptl t38\r\n");
 	}
 
 	/* Update lastrtprx when we send our SDP */
@@ -9428,6 +9470,8 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
 
 	if (p->do_history)
 		append_history(p, "ReInv", "Re-invite sent");
+	memset(p->offered_media, 0, sizeof(p->offered_media));
+
 	if (t38version)
 		add_sdp(&req, p, oldsdp, FALSE, TRUE);
 	else
@@ -9833,6 +9877,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
 		ast_channel_unlock(chan);
 	}
 	if (sdp) {
+		memset(p->offered_media, 0, sizeof(p->offered_media));
 		if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
 			ast_udptl_offered_from_local(p->udptl, 1);
 			ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");