[asterisk-commits] branch oej/sdpcleanup r26886 - in /team/oej/sdpcleanup: ./ channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu May 11 06:57:42 MST 2006


Author: oej
Date: Thu May 11 08:57:41 2006
New Revision: 26886

URL: http://svn.digium.com/view/asterisk?rev=26886&view=rev
Log:
Update

Modified:
    team/oej/sdpcleanup/   (props changed)
    team/oej/sdpcleanup/channels/chan_sip.c

Propchange: team/oej/sdpcleanup/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Modified: team/oej/sdpcleanup/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/channels/chan_sip.c?rev=26886&r1=26885&r2=26886&view=diff
==============================================================================
--- team/oej/sdpcleanup/channels/chan_sip.c (original)
+++ team/oej/sdpcleanup/channels/chan_sip.c Thu May 11 08:57:41 2006
@@ -792,7 +792,7 @@
 	int jointcapability;			/*!< Supported capability at both ends (codecs ) */
 	int peercapability;			/*!< Supported peer capability */
 	int prefcodec;				/*!< Preferred codec (outbound only) */
-	int noncodeccapability;
+	int noncodeccapability;			/*!< RTP non-codec capabilities */
 	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
 	int callingpres;			/*!< Calling presentation */
 	int authtries;				/*!< Times we've tried to authenticate */
@@ -1236,6 +1236,10 @@
 static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan);
 static int sip_get_codec(struct ast_channel *chan);
 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p);
+
+/*----- SDP handling functions */
+static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
+static int process_sdp(struct sip_pvt *p, struct sip_request *req);
 
 /*! \brief Definition of this channel for PBX channel registration */
 static const struct ast_channel_tech sip_tech = {
@@ -3156,6 +3160,7 @@
 		return NULL;
 	}
 	tmp->tech = &sip_tech;
+
 	/* Select our native format based on codec preference until we receive
 	   something from another device to the contrary. */
 	if (i->jointcapability)
@@ -3164,11 +3169,22 @@
 		what = i->capability;
 	else
 		what = global_capability;
-	needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK;
-	if (option_debug && needvideo) 
-		ast_log(LOG_DEBUG, "This call needs video! HOLLYWOOD next!\n");
+
+	needvideo = what & AST_FORMAT_VIDEO_MASK;
+
+	if (option_debug) {
+		if (needvideo) 
+			ast_log(LOG_DEBUG, "This call needs video! HOLLYWOOD next!\n");
+		else
+			ast_log(LOG_DEBUG, "This call does not need video.\n");
+	}
 
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
+	if (option_debug) {
+		char buf[BUFSIZ];
+		ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
+	}
+
 	fmt = ast_best_codec(tmp->nativeformats);
 
 	if (title)
@@ -3852,8 +3868,8 @@
 	int portno = -1;		/*!< RTP Audio port number */
 	int vportno = -1;		/*!< RTP Video port number */
 	int peercapability, peernoncodeccapability;
-	int vpeercapability=0, vpeernoncodeccapability=0;
-	struct sockaddr_in sin;
+	int vpeercapability = 0, vpeernoncodeccapability = 0;
+	struct sockaddr_in sin;		/*!< media socket address */
 	struct sockaddr_in vsin;	/*!< Video socket address */
 	const char *codecs;
 	struct hostent *hp;		/*!< RTP Audio host IP */
@@ -3879,10 +3895,9 @@
 		return -1;
 	}
 
+	/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
 	ast_rtp_pt_default(&newaudiortp);
-	if (p->vrtp)
-		ast_rtp_pt_default(&newvideortp);
-
+	ast_rtp_pt_default(&newvideortp);
 
 	/* Update our last rtprx when we receive an SDP, too */
 	time(&p->lastrtprx);
@@ -3893,6 +3908,7 @@
 		ast_log(LOG_NOTICE, "Content is '%s', not 'application/sdp'\n", get_header(req, "Content-Type"));
 		return -1;
 	}
+
 	/* Try to find first media stream */
 	m = get_sdp(req, "m");
 	destiterator = 0;
@@ -3901,16 +3917,21 @@
 		ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
 		return -1;
 	}
+
+	/* Check for IPv4 address (not IPv6 yet) */
 	if (sscanf(c, "IN IP4 %256s", host) != 1) {
 		ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
 		return -1;
 	}
+
 	/* XXX This could block for a long time, and block the main thread! XXX */
 	hp = ast_gethostbyname(host, &audiohp);
 	if (!hp) {
 		ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
 		return -1;
 	}
+	vhp = hp;	/* Copy to video address as default too */
+	
 	iterator = 0;
 	ast_set_flag(&p->flags[0], SIP_NOVIDEO);	
 
@@ -3998,19 +4019,6 @@
 	sin.sin_port = htons(portno);
 	/* Setup video port number */
 	vsin.sin_port = htons(vportno);
-
-	if (p->rtp && sin.sin_port) {
-		ast_rtp_set_peer(p->rtp, &sin);
-		if (debug)
-			ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
-	}
-
-	/* Setup video port number */
-	if (p->vrtp && vsin.sin_port) {
-		ast_rtp_set_peer(p->vrtp, &vsin);
-		if (debug) 
-			ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), vsin.sin_addr), ntohs(vsin.sin_port));
-	}
 
 	/* Next, scan through each "a=rtpmap:" line, noting each
 	 * specified RTP payload type (with corresponding MIME subtype):
@@ -4066,8 +4074,7 @@
 
 	/* Now gather all of the codecs that we are asked for: */
 	ast_rtp_get_current_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
-	if (p->vrtp)
-		ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
+	ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
 
 	newjointcapability = p->capability & (peercapability | vpeercapability);
 	newpeercapability = (peercapability | vpeercapability);
@@ -4106,19 +4113,12 @@
 		return -1;
 	}
 
-	if (!p->owner) 	/* There's no open channel owning us */
-		return 0;
-
-	/* Ok, we're going with this offer */
-	if (debug && option_debug)
-		ast_log(LOG_DEBUG, "Ok, we're accepting this SDP offer! \n");
-
+	/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
+		they are acceptable */
 	p->jointcapability = newjointcapability;
 	p->peercapability = newpeercapability;
 	p->noncodeccapability = newnoncodeccapability;
 		
-	/* We are now ready to change p->rtp and p->vrtp with the offered codecs, since
-		they are acceptable */
 	{
 		int i;
 		/* Copy payload types from source to destination */
@@ -4131,6 +4131,30 @@
 			}
 		}
 	}
+
+	/* Setup audio port number */
+	if (p->rtp && sin.sin_port) {
+		ast_rtp_set_peer(p->rtp, &sin);
+		if (debug)
+			ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+	}
+
+	/* Setup video port number */
+	if (p->vrtp && vsin.sin_port) {
+		ast_rtp_set_peer(p->vrtp, &vsin);
+		if (debug) 
+			ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), vsin.sin_addr), ntohs(vsin.sin_port));
+	}
+
+	/* Ok, we're going with this offer */
+	if (option_debug) {
+		char buf[BUFSIZ];
+		ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->capability));
+	}
+
+	if (!p->owner) 	/* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
+		return 0;
+
 
 	if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
 		const unsigned slen=512;
@@ -4192,6 +4216,7 @@
 		}
 		ast_set_flag(&p->flags[0], SIP_CALL_ONHOLD);
 	}
+	
 
 	return 0;
 }
@@ -4847,7 +4872,7 @@
 	int alreadysent = 0;
 	struct sockaddr_in sin;
 	struct sockaddr_in vsin;
-	char v[256];
+	char v[10];
 	char s[256];
 	char o[256];
 	char c[256];
@@ -4871,6 +4896,7 @@
 	struct sockaddr_in dest;
 	struct sockaddr_in vdest = { 0, };
 	int debug;
+	int needvideo = FALSE;
 	
 	debug = sip_debug_test_pvt(p);
 
@@ -4880,7 +4906,7 @@
 		return -1;
 	}
 	capability = p->jointcapability;
-		
+
 	if (!p->sessionid) {
 		p->sessionid = getpid();
 		p->sessionversion = p->sessionid;
@@ -4890,6 +4916,7 @@
 	if (p->vrtp)
 		ast_rtp_get_us(p->vrtp, &vsin);
 
+	/* Is this a re-invite to move the media out, then use the original offer from caller  */
 	if (p->redirip.sin_addr.s_addr) {
 		dest.sin_port = p->redirip.sin_port;
 		dest.sin_addr = p->redirip.sin_addr;
@@ -4900,8 +4927,18 @@
 		dest.sin_port = sin.sin_port;
 	}
 
+	if(capability & AST_FORMAT_VIDEO_MASK) {
+		if (p->vrtp) {
+			needvideo = TRUE;
+			if (option_debug)
+				ast_log(LOG_DEBUG, "This call needs video offers! \n");
+		} else if (option_debug)
+			ast_log(LOG_DEBUG, "This call needs video offers, but there's no video suupport enabled ! \n");
+	}
+		
+
 	/* Determine video destination */
-	if (p->vrtp) {
+	if (needvideo) {
 		if (p->vredirip.sin_addr.s_addr) {
 			vdest.sin_port = p->vredirip.sin_port;
 			vdest.sin_addr = p->vredirip.sin_addr;
@@ -4912,7 +4949,7 @@
 	}
 	if (debug) {
 		ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));	
-		if (p->vrtp)
+		if (needvideo)
 			ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port));	
 	}
 
@@ -4939,7 +4976,7 @@
 					 &m_audio_next, &m_audio_left,
 					 &a_audio_next, &a_audio_left,
 					 debug);
-		else
+		else if (needvideo)
 			add_codec_to_sdp(p, p->prefcodec, 90000,
 					 &m_video_next, &m_video_left,
 					 &a_video_next, &a_video_left,
@@ -4963,7 +5000,7 @@
 					 &m_audio_next, &m_audio_left,
 					 &a_audio_next, &a_audio_left,
 					 debug);
-		else
+		else if (needvideo)
 			add_codec_to_sdp(p, pref_codec, 90000,
 					 &m_video_next, &m_video_left,
 					 &a_video_next, &a_video_left,
@@ -5024,7 +5061,7 @@
 	add_line(resp, o);
 	add_line(resp, s);
 	add_line(resp, c);
-	if ((p->vrtp) &&
+	if ((needvideo) &&
 	    (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
 	    (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
 		add_line(resp, b);
@@ -5041,6 +5078,11 @@
 	/* Update lastrtprx when we send our SDP */
 	time(&p->lastrtprx);
 	time(&p->lastrtptx);
+
+	if (option_debug) {
+		char buf[BUFSIZ];
+		ast_log(LOG_DEBUG, "We're settling with offering these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
+	}
 
 	return 0;
 }
@@ -7610,20 +7652,24 @@
 			p->callgroup = user->callgroup;
 			p->pickupgroup = user->pickupgroup;
 			p->callingpres = user->callingpres;
-			p->capability = user->capability;
-			p->jointcapability = user->capability;
-			p->maxcallbitrate = user->maxcallbitrate;
-			if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
-				ast_rtp_destroy(p->vrtp);
-				p->vrtp = NULL;
-			}
-			if (p->peercapability)
+			
+			/* Set default codec settings for this call */
+			p->capability = user->capability;		/* User codec choice */
+			p->jointcapability = user->capability;		/* Our codecs */
+			if (p->peercapability)				/* AND with peer's codecs */
 				p->jointcapability &= p->peercapability;
 			if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
 			    (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
 				p->noncodeccapability |= AST_RTP_DTMF;
 			else
 				p->noncodeccapability &= ~AST_RTP_DTMF;
+
+			p->maxcallbitrate = user->maxcallbitrate;
+			/* If we do not support video, remove video from call structure */
+			if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
+				ast_rtp_destroy(p->vrtp);
+				p->vrtp = NULL;
+			}
 		}
 		if (user && debug)
 			ast_verbose("Found user '%s'\n", user->name);
@@ -10092,10 +10138,11 @@
 		if (!ast_test_flag(req, SIP_PKT_IGNORE))
 			sip_cancel_destroy(p);
 		p->authtries = 0;
-		if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) 
-			if (res = process_sdp(p, req) && !ast_test_flag(req, SIP_PKT_IGNORE))
+		if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
+			if ((res = process_sdp(p, req)) && !ast_test_flag(req, SIP_PKT_IGNORE))
 				/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
 				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
+		}
 
 		/* Parse contact header for continued conversation */
 		/* When we get 200 OK, we know which device (and IP) to contact for this call */
@@ -12610,7 +12657,7 @@
 #if 0
 	printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
 #endif
-	p->prefcodec = format;
+	p->prefcodec = format;				/* Format for this call */
 	ast_mutex_lock(&p->lock);
 	tmpc = sip_new(p, AST_STATE_DOWN, host);	/* Place the call */
 	ast_mutex_unlock(&p->lock);
@@ -12912,6 +12959,7 @@
 	ast_copy_flags(&user->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
 	user->capability = global_capability;
 	user->allowtransfer = global_allowtransfer;
+	user->maxcallbitrate = default_maxcallbitrate;
 	user->prefs = default_prefs;
 	/* set default context */
 	strcpy(user->context, default_context);
@@ -13800,8 +13848,9 @@
 	}
 	if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
 		if (chan->_state != AST_STATE_UP) {
-				char iabuf[INET_ADDRSTRLEN];
-				ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
+			char iabuf[INET_ADDRSTRLEN];
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
 		} else if (!p->pendinginvite) {
 			if (option_debug > 2) {
 				char iabuf[INET_ADDRSTRLEN];



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