Index: channels/chan_sip.c
===================================================================
--- ../asterisk-1.4.4/channels/chan_sip.c	2007-04-27 16:04:07.000000000 +0200
+++ channels/chan_sip.c	2007-05-27 19:26:45.000000000 +0200
@@ -1210,6 +1211,7 @@
 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
 static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
+static int sip_setoption(struct ast_channel *c, int option, void *data, int datalen);
 
 /*--- Transmitting responses and requests */
 static int sipsock_read(int *id, int fd, short events, void *ignore);
@@ -1543,11 +1546,7 @@
 	.write_video = sip_write,
 	.indicate = sip_indicate,
 	.transfer = sip_transfer,
+	.setoption = sip_setoption,
 	.fixup = sip_fixup,
 	.send_digit_begin = sip_senddigit_begin,
 	.send_digit_end = sip_senddigit_end,
@@ -3456,9 +3480,29 @@
 	struct sip_pvt *p = ast->tech_pvt;
 
 	ast_mutex_lock(&p->lock);
-	if (ast->_state != AST_STATE_UP) {
+	if (ast->_state != AST_STATE_UP)
+	{
 		try_suggested_sip_codec(p);	
 
+		/* new mechanism based on prefcodecs to negotiate SDP 
+                 * in answers */
+		if (p->prefcodec)
+		{
+			int afmt, vfmt;
+
+			afmt = p->jointcapability & p->prefcodec & AST_FORMAT_AUDIO_MASK;
+			vfmt = p->jointcapability & p->prefcodec & AST_FORMAT_VIDEO_MASK;
+			if (afmt == 0)
+			{
+				 ast_log(LOG_NOTICE, "No common audio codec with preferred codec, changing only video part to %08x. Audio is transcoded.\n", vfmt);
+				 p->jointcapability &= (vfmt | AST_FORMAT_AUDIO_MASK);
+			}
+			else
+			{
+				ast_log(LOG_NOTICE, "Adjusting jointcapabilities based on preferred codec format %08x.\n", p->prefcodec);
+				 p->jointcapability &= p->prefcodec;
+			}
+		}
 		ast_setstate(ast, AST_STATE_UP);
 		if (option_debug)
 			ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
@@ -3656,6 +3700,47 @@
 	return res;
 }
 
+/*  SIP set option. Mostly indicate the preferred codec to this channel.
+ * this is useful for app_dial and app_queue to close the codec negocialtion
+ * and send back a proper 200 OK or 183 Call Progress with a negociated SDP
+ * instead using capapbility to create the SDP */
+static int sip_setoption(struct ast_channel *c, int option, void *data, int datalen)
+{
+	struct sip_pvt *p = c->tech_pvt;
+	int * p_fmt, vfmt;
+
+	switch (option)
+	{
+		case AST_OPTION_PREFCODECS:
+			if (datalen != sizeof(int))
+			{
+				ast_log(LOG_WARNING,"chan %s - AST_OPTION_PREFCODEC data len incorrect. Ignoring.\n", c->name);
+				return -2;
+			}
+
+			p_fmt = (int *) data;
+			/* check video compatibility */
+			vfmt = (*p_fmt) & AST_FORMAT_VIDEO_MASK;
+			if ( vfmt != 0 
+			     &&
+			     (vfmt & c->nativeformats & AST_FORMAT_VIDEO_MASK) == 0)
+			{
+				ast_log(LOG_WARNING,"chan %s - video format requested %08x is incompatble with channel native video format %08x.\n", c->name, vfmt, c->nativeformats & AST_FORMAT_VIDEO_MASK);
+				return -3;
+			}
+			ast_log(LOG_DEBUG,"chan %s - setting preferred codecs to %08x.\n", c->name, *p_fmt);
+
+			p->prefcodec = *p_fmt;
+			break;
+
+				
+		default:
+			ast_log(LOG_NOTICE, "chan %s - unsupported option %d rquested.\n", c->name, option);
+		return -1;
+	}
+	return 0;
+}
+
 /*! \brief Play indication to user 
  * With SIP a lot of indications is sent as messages, letting the device play
    the indication - busy signal, congestion etc 
@@ -5222,14 +5307,19 @@
 	if (option_debug > 3)
 		ast_log(LOG_DEBUG, "We have an owner, now see if we need to change this call\n");
 
-	if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
-		if (debug) {
-			char s1[BUFSIZ], s2[BUFSIZ];
-			ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", 
-				ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
-				ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
-		}
+	/* if peer and incoming channel have no audio codec in common, 
+           pickup new codecs to prepare translation
+         */
+	if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)
+		&& 
+	    (p->jointcapability & AST_FORMAT_AUDIO_MASK))
+	{
+		char s1[BUFSIZ], s2[BUFSIZ];
+		ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", 
+			ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
+			ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
 		p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
+			ast_log(LOG_DEBUG, "New native format for chan %s is %08x\n", p->owner->name,  p->owner->nativeformats);
 		ast_set_read_format(p->owner, p->owner->readformat);
 		ast_set_write_format(p->owner, p->owner->writeformat);
 	}
@@ -15440,6 +15599,7 @@
 	printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
 #endif
 	p->prefcodec = oldformat;				/* Format for this call */
+	p->jointcapability = oldformat & p->capability;
 	ast_mutex_lock(&p->lock);
 	tmpc = sip_new(p, AST_STATE_DOWN, host);	/* Place the call */
 	ast_mutex_unlock(&p->lock);
Index: main/channel.c
===================================================================
--- ../asterisk-1.4.4/main/channel.c	2007-04-26 05:19:51.000000000 +0200
+++ main/channel.c	2007-05-27 18:56:06.000000000 +0200
@@ -2863,13 +2863,16 @@
 static int set_format(struct ast_channel *chan, int fmt, int *rawformat, int *format,
 		      struct ast_trans_pvt **trans, const int direction)
 {
-	int native;
+	int native, oldfmt;
 	int res;
-	
+
+        oldfmt = fmt;	
 	/* Make sure we only consider audio */
 	fmt &= AST_FORMAT_AUDIO_MASK;
 	
-	native = chan->nativeformats;
+	//native = AST_FORMAT_AUDIO_MASK & chan->nativeformats;
+	native =  chan->nativeformats;
+        ast_log(LOG_DEBUG, "chan=%s format=%08x (with video = %08x) native=%08x in dir. %s\n", chan->name, fmt, oldfmt, native, direction ? "write" : "read");
 	/* Find a translation path from the native format to one of the desired formats */
 	if (!direction)
 		/* reading */
@@ -3081,7 +3084,7 @@
 		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);
+			ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %08x) to %08x\n", type, chan->tech->capabilities, format);
 			AST_LIST_UNLOCK(&channels);
 			return NULL;
 		}
@@ -3212,55 +3215,116 @@
 
 int ast_channel_make_compatible(struct ast_channel *chan, struct ast_channel *peer)
 {
-	int src;
-	int dst;
+	int asrc, vsrc;
+	int adst, vdst;
+	
+        ast_log(LOG_DEBUG, "src chan %s with format %08x to dst chan %s with format %08x\n", chan->name,  chan->nativeformats, peer->name, peer->nativeformats);
 
-	/* Set up translation from the chan to the peer */
-	src = chan->nativeformats;
-	dst = peer->nativeformats;
-	if (ast_translator_best_choice(&dst, &src) < 0) {
-		ast_log(LOG_WARNING, "No path to translate from %s(%d) to %s(%d)\n", chan->name, src, peer->name, dst);
+	/* Set up AUDIO translation from the chan to the peer */
+	asrc = chan->nativeformats & AST_FORMAT_AUDIO_MASK;
+	adst = peer->nativeformats & AST_FORMAT_AUDIO_MASK;
+	if (ast_translator_best_choice(&adst, &asrc) < 0) {
+		ast_log(LOG_WARNING, "No path to translate AUDIO from chan=%s (%08x) to peer=%s (%08x)\n", chan->name, asrc, peer->name, adst);
 		return -1;
 	}
 
+	ast_log(LOG_DEBUG, "ast_translator_best_choice() selected AUDIO src format  %08x and AUDIO dst format  %08x\n", asrc, adst);
 	/* if the best path is not 'pass through', then
 	   transcoding is needed; if desired, force transcode path
 	   to use SLINEAR between channels, but only if there is
 	   no direct conversion available */
-	if ((src != dst) && ast_opt_transcode_via_slin &&
-	    (ast_translate_path_steps(dst, src) != 1))
-		dst = AST_FORMAT_SLINEAR;
-	if (ast_set_read_format(chan, dst) < 0) {
-		ast_log(LOG_WARNING, "Unable to set read format on channel %s to %d\n", chan->name, dst);
-		return -1;
-	}
-	if (ast_set_write_format(peer, dst) < 0) {
-		ast_log(LOG_WARNING, "Unable to set write format on channel %s to %d\n", peer->name, dst);
-		return -1;
-	}
-
-	/* Set up translation from the peer to the chan */
-	src = peer->nativeformats;
-	dst = chan->nativeformats;
-	if (ast_translator_best_choice(&dst, &src) < 0) {
-		ast_log(LOG_WARNING, "No path to translate from %s(%d) to %s(%d)\n", peer->name, src, chan->name, dst);
-		return -1;
+	if ((asrc != adst)
+            &&
+            ast_opt_transcode_via_slin
+            &&
+	    (ast_translate_path_steps(adst, asrc) != 1))
+	{
+		ast_log(LOG_NOTICE, "No direct AUDIO transl -> transcoding audio using SLINEAR format between channels\n");
+		adst = AST_FORMAT_SLINEAR;
+	}
+
+        /* Set up VIDEO translation from the chan to the peer */
+        vsrc = chan->nativeformats & AST_FORMAT_VIDEO_MASK;
+        vdst = peer->nativeformats & AST_FORMAT_VIDEO_MASK;
+        if (ast_translator_best_choice(&vdst, &vsrc) < 0) {
+                ast_log(LOG_WARNING, "No path to translate VIDEO from chan=%s(%08x) to peer=%s(%08x)\n", chan->name, vsrc, peer->name, vdst);
+                return -2;
+        }
+
+        ast_log(LOG_DEBUG, "after ast_translator_best_choice(), VIDEO src format is %08x and VIDEO dst format is %08x\n", vsrc, vdst);
+
+        if ( vsrc != vdst && ast_translate_path_steps(vdst, vsrc) != 1)
+        {
+                ast_log(LOG_ERROR, "No direct VIDEO translation. Failed to make channels compatible");
+                return -2;
+        }
+
+        if (ast_set_read_format(chan, adst | vdst) < 0) {
+                ast_log(LOG_WARNING, "Unable to set read format on channel %s to %08x\n", chan->name, adst | vdst);
+                return -3;
+        }
+
+        if (ast_set_write_format(peer, adst | vdst) < 0) {
+                ast_log(LOG_WARNING, "Unable to set write format on channel %s to %d\n", peer->name, adst | vdst);
+                return -4;
+        }
+
+	/* Set up translation from the PEER to the CHAN for AUDIO part*/
+	asrc = peer->nativeformats & AST_FORMAT_AUDIO_MASK;
+	adst = chan->nativeformats & AST_FORMAT_AUDIO_MASK;
+	if (ast_translator_best_choice(&adst, &asrc) < 0) {
+		ast_log(LOG_WARNING, "No path to translate AUDIO from peer=%s(%08x) to chan=%s(%08x)\n", peer->name, asrc, chan->name, adst);
+		return -2;
 	}
 
 	/* if the best path is not 'pass through', then
 	   transcoding is needed; if desired, force transcode path
 	   to use SLINEAR between channels, but only if there is
 	   no direct conversion available */
-	if ((src != dst) && ast_opt_transcode_via_slin &&
-	    (ast_translate_path_steps(dst, src) != 1))
-		dst = AST_FORMAT_SLINEAR;
-	if (ast_set_read_format(peer, dst) < 0) {
-		ast_log(LOG_WARNING, "Unable to set read format on channel %s to %d\n", peer->name, dst);
-		return -1;
+	if ((asrc != adst)
+		&&
+	     ast_opt_transcode_via_slin
+		&&
+	    (ast_translate_path_steps(adst, asrc) != 1))
+	{
+		ast_log(LOG_NOTICE, "No direct AUDIO transl -> transcoding audio using SLINEAR format between channels\n");
+
 	}
-	if (ast_set_write_format(chan, dst) < 0) {
-		ast_log(LOG_WARNING, "Unable to set write format on channel %s to %d\n", chan->name, dst);
-		return -1;
+
+	/* Set up translation from the PEER to the CHAN for VIDEO part*/
+        vsrc = peer->nativeformats & AST_FORMAT_VIDEO_MASK;
+        vdst = chan->nativeformats & AST_FORMAT_VIDEO_MASK;
+        if (ast_translator_best_choice(&vdst, &vsrc) < 0) {
+                ast_log(LOG_WARNING, "No path to translate VIDEO from peer=%s(%08x) to chan=%s(%08x)\n", peer->name, asrc, chan->name, adst);
+                return -5;
+        }
+
+        ast_log(LOG_DEBUG, "after ast_translator_best_choice(), VIDEO src format is %08x and AUDIO dst format is %08x\n", vsrc, vdst);
+
+        if ( vsrc != vdst && ast_translate_path_steps(vdst, vsrc) != 1)
+        {
+                ast_log(LOG_ERROR, "No direct VIDEO translation. Failed to make channels compatible");
+                return -5;
+        }
+
+	if (ast_set_read_format(peer, adst | vdst) < 0) {
+		ast_log(LOG_WARNING, "Unable to set read format on peer channel %s to %d\n", peer->name, adst | vdst);
+		return -6;
+	}
+	if (ast_set_write_format(chan, adst | vdst) < 0) {
+		ast_log(LOG_WARNING, "Unable to set write format on channel %s to %d\n", chan->name, adst | vdst);
+		return -7;
+	}
+
+	/* new ! indicate preferred codec to incoming channel. Restrict here to
+           SIP channels because it requeires the siport of a new AST_OPTION but
+	   the same mechanism could be used for IAX and later for H.324m or 
+           other protocols which supports codec nego (skinny ? gtalk ?)
+	 */
+	if ( !strncmp(chan->name,"SIP",3) )
+	{
+		int pcodec = adst | vdst;
+		ast_channel_setoption(chan, AST_OPTION_PREFCODECS, &pcodec, sizeof(pcodec), 0);
 	}
 	return 0;
 }
Index: include/asterisk/frame.h
===================================================================
--- ../asterisk-1.4.4/include/asterisk/frame.h	2006-10-30 17:27:34.000000000 +0100
+++ include/asterisk/frame.h	2007-05-27 18:01:05.000000000 +0200
@@ -329,6 +331,8 @@
 /*! Explicitly enable or disable echo cancelation for the given channel */
 #define	AST_OPTION_ECHOCAN		8
 
+/*! indicate the given channel what are our preferred codecs */
+#define	AST_OPTION_PREFCODECS		9
 struct oprmode {
 	struct ast_channel *peer;
 	int mode;