Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests

and some patches (all disclaimed).

- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments

Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channel.c

@@ -539,7 +539,7 @@ char *ast_transfercapability2str(int transfercapability)
 	}
 }
 
-/*! \brief Pick the best codec */
+/*! \brief Pick the best audio codec */
 int ast_best_codec(int fmts)
 {
 	/* This just our opinion, expressed in code.  We are asked to choose
@@ -573,6 +573,8 @@ int ast_best_codec(int fmts)
 		AST_FORMAT_G723_1,
 	};
 
+	/* Strip out video */
+	fmts &= AST_FORMAT_AUDIO_MASK;
 	
 	/* Find the first preferred codec in the format given */
 	for (x=0; x < (sizeof(prefs) / sizeof(prefs[0]) ); x++)
@@ -2614,6 +2616,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
 	int fmt;
 	int res;
 	int foo;
+	int videoformat = format & AST_FORMAT_VIDEO_MASK;
 
 	if (!cause)
 		cause = &foo;
@@ -2629,7 +2632,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
 			continue;
 
 		capabilities = chan->tech->capabilities;
-		fmt = format;
+		fmt = format & AST_FORMAT_AUDIO_MASK;
 		res = ast_translator_best_choice(&fmt, &capabilities);
 		if (res < 0) {
 			ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %d) to %d\n", type, chan->tech->capabilities, format);
@@ -2640,7 +2643,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
 		if (!chan->tech->requester)
 			return NULL;
 		
-		if (!(c = chan->tech->requester(type, capabilities, data, cause)))
+		if (!(c = chan->tech->requester(type, capabilities | videoformat, data, cause)))
 			return NULL;
 
 		if (c->_state == AST_STATE_DOWN) {

frame.c

@@ -1008,9 +1008,12 @@ int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best)
 			break;
 		}
 	}
-	if(ret)
+	if(ret & AST_FORMAT_AUDIO_MASK)
 		return ret;
 
+	if (option_debug > 3)
+		ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
+
    	return find_best ? ast_best_codec(formats) : 0;
 }
 
@@ -1034,7 +1037,10 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char
 				*mask &= ~format;
 		}
 
-		if (pref) {
+		/* Set up a preference list for audio. Do not include video in preferences 
+		   since we can not transcode video and have to use whatever is offered
+		 */
+		if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
 			if (strcasecmp(this, "all")) {
 				if (allowing)
 					ast_codec_pref_append(pref, format);

include/asterisk/rtp.h

@@ -40,13 +40,13 @@ extern "C" {
 
 /* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
 /*! DTMF (RFC2833) */
-#define AST_RTP_DTMF            (1 << 0)
+#define AST_RTP_DTMF            	(1 << 0)
 /*! 'Comfort Noise' (RFC3389) */
-#define AST_RTP_CN              (1 << 1)
+#define AST_RTP_CN              	(1 << 1)
 /*! DTMF (Cisco Proprietary) */
-#define AST_RTP_CISCO_DTMF      (1 << 2)
+#define AST_RTP_CISCO_DTMF      	(1 << 2)
 /*! Maximum RTP-specific code */
-#define AST_RTP_MAX             AST_RTP_CISCO_DTMF
+#define AST_RTP_MAX             	AST_RTP_CISCO_DTMF
 
 #define MAX_RTP_PT			256
 
@@ -62,6 +62,9 @@ struct ast_rtp_protocol {
 	AST_LIST_ENTRY(ast_rtp_protocol) list;
 };
 
+
+#define FLAG_3389_WARNING		(1 << 0)
+
 typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
 
 
@@ -71,7 +74,6 @@ typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *
  * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP.  A participant may be involved in multiple RTP sessions at the same time [...]"
  *
  */
-
 /*! \brief The value of each payload format mapping: */
 struct rtpPayloadType {
 	int isAstFormat; 	/*!< whether the following code is an AST_FORMAT */

include/asterisk/translate.h

@@ -23,7 +23,8 @@
 #ifndef _ASTERISK_TRANSLATE_H
 #define _ASTERISK_TRANSLATE_H
 
-#define MAX_FORMAT 32
+//#define MAX_FORMAT 15	/* Do not include video here */
+#define MAX_FORMAT 32	/* Do include video here */
 
 #if defined(__cplusplus) || defined(c_plusplus)
 extern "C" {

rtp.c

@@ -1350,7 +1350,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
 	return 1;
 }
 
-/*! \brief  Make a note of a RTP paymoad type that was seen in a SDP "m=" line.
+/*! \brief  Make a note of a RTP payload type that was seen in a SDP "m=" line.
  * By default, use the well-known value for this type (although it may 
  * still be set to a different value by a subsequent "a=rtpmap:" line)
  */
@@ -1359,9 +1359,8 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
 	if (pt < 0 || pt > MAX_RTP_PT) 
 		return; /* bogus payload type */
 
-	if (static_RTP_PT[pt].code != 0) {
+	if (static_RTP_PT[pt].code != 0) 
 		rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
-	}
 } 
 
 /*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
@@ -2245,7 +2244,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
 	
 	/* Make sure we have enough space for RTP header */
 	if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
-		ast_log(LOG_WARNING, "RTP can only send voice\n");
+		ast_log(LOG_WARNING, "RTP can only send voice and video\n");
 		return -1;
 	}