--- asterisk-1.4.17/configs/sip.conf.sample	2007-11-27 08:34:19.000000000 +0100
+++ asterisk-1.4.17-h324m/configs/sip.conf.sample	2008-03-04 19:57:54.000000000 +0100
@@ -123,6 +123,12 @@
 ;maxcallbitrate=384		; Maximum bitrate for video calls (default 384 kb/s)
 				; Videosupport and maxcallbitrate is settable
 				; for peers and users as well
+;maxcallbitrate_h263p=80	; Maximum bitrate for h263p video calls. 0 means 'no maximum'
+				; thus no header is added to SDP. (default 0 kb/s)
+				; This is only a global setting. This option uses the obsolete
+				; MaxBR parameter. Successful tested with eyebeam.
+				; 80 kbit is fine for SIP<-->H324M calls
+				; it also adds the standard conform b=TIAS:..... parameter
 ;callevents=no			; generate manager events when sip ua 
 				; performs events (e.g. hold)
 ;alwaysauthreject = yes		; When an incoming INVITE or REGISTER is to be rejected,
--- asterisk-1.4.17/channels/chan_sip.c	2008-01-02 21:24:09.000000000 +0100
+++ asterisk-1.4.17-h324m/channels/chan_sip.c	2008-01-18 12:01:26.000000000 +0100
@@ -510,6 +510,7 @@
 #define DEFAULT_QUALIFY		FALSE
 #define DEFAULT_T1MIN		100		/*!< 100 MS for minimal roundtrip time */
 #define DEFAULT_MAX_CALL_BITRATE (384)		/*!< Max bitrate for video */
+#define DEFAULT_MAX_CALL_BITRATE_H263P (0)	/*!< Max bitrate for video for h263p codec */
 #ifndef DEFAULT_USERAGENT
 #define DEFAULT_USERAGENT "Asterisk PBX"	/*!< Default Useragent: header unless re-defined in sip.conf */
 #endif
@@ -528,7 +529,8 @@
 static char default_mohinterpret[MAX_MUSICCLASS];  /*!< Global setting for moh class to use when put on hold */
 static char default_mohsuggest[MAX_MUSICCLASS];	   /*!< Global setting for moh class to suggest when putting 
                                                     *   a bridged channel on hold */
-static int default_maxcallbitrate;	/*!< Maximum bitrate for call */
+static int default_maxcallbitrate;		/*!< Maximum bitrate for call */
+static int default_maxcallbitrate_h263p;	/*!< Maximum bitrate for call for h263p codec */
 static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
 
 /* Global settings only apply to the channel */
@@ -960,6 +962,7 @@
 	int jointnoncodeccapability;            /*!< Joint Non codec capability */
 	int redircodecs;			/*!< Redirect codecs */
 	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
+	int maxcallbitrate_h263p;		/*!< Maximum Call Bitrate for Video Calls with h263p codec*/	
 	struct t38properties t38;		/*!< T38 settings */
 	struct sockaddr_in udptlredirip;	/*!< Where our T.38 UDPTL should be going if not to us */
 	struct ast_udptl *udptl;		/*!< T.38 UDPTL session */
@@ -4463,6 +4466,7 @@
 		if (p->udptl)
 			ast_udptl_settos(p->udptl, global_tos_audio);
 		p->maxcallbitrate = default_maxcallbitrate;
+		p->maxcallbitrate_h263p = default_maxcallbitrate_h263p;
 	}
 
 	if (useglobal_nat && sin) {
@@ -6180,6 +6184,10 @@
 	} else if (codec == AST_FORMAT_ILBC) {
 		/* Add information about us using only 20/30 ms packetization */
 		ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+	} else if (codec == AST_FORMAT_H263_PLUS) {
+		/* Add eyebeam style maxbandwidth description */
+		if (p->maxcallbitrate_h263p)
+			ast_build_string(a_buf, a_size, "a=fmtp:%d QCIF=1 MaxBR=%d0\r\n", rtp_code,p->maxcallbitrate_h263p);
 	}
 
 	if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
@@ -6367,12 +6375,15 @@
 	char *hold;
 	char m_audio[256];				/* Media declaration line for audio */
 	char m_video[256];				/* Media declaration line for video */
+	char b_video[1024];				/* Bandwidth for video */
 	char a_audio[1024];				/* Attributes for audio */
 	char a_video[1024];				/* Attributes for video */
 	char *m_audio_next = m_audio;
 	char *m_video_next = m_video;
 	size_t m_audio_left = sizeof(m_audio);
 	size_t m_video_left = sizeof(m_video);
+	char *b_video_next = b_video;
+	size_t b_video_left = sizeof(b_video);
 	char *a_audio_next = a_audio;
 	char *a_video_next = a_video;
 	size_t a_audio_left = sizeof(a_audio);
@@ -6538,6 +6549,10 @@
 					 debug, &min_video_packet_size);
 	}
 
+	/* Build b= bandwidth SDP lines for video */
+	if ((needvideo) && (p->maxcallbitrate_h263p))
+		ast_build_string(&b_video_next, &b_video_left, "b=TIAS:%d\r\n", (p->maxcallbitrate_h263p)*1000);
+
 	/* Now add DTMF RFC2833 telephony-event as a codec */
 	for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
 		if (!(p->jointnoncodeccapability & x))
@@ -6570,7 +6585,7 @@
 
 	len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
 	if (needvideo) /* only if video response is appropriate */
-		len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold);
+		len += strlen(m_video) + strlen(b_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold);
 
 	add_header(resp, "Content-Type", "application/sdp");
 	add_header_contentLength(resp, len);
@@ -6586,6 +6601,7 @@
 	add_line(resp, hold);
 	if (needvideo) { /* only if video response is appropriate */
 		add_line(resp, m_video);
+		add_line(resp, b_video);
 		add_line(resp, a_video);
 		add_line(resp, hold);	/* Repeat hold for the video stream */
 	}
@@ -10634,6 +10650,7 @@
 	ast_cli(fd, "  Notify hold state:      %s\n", global_notifyhold ? "Yes" : "No");
 	ast_cli(fd, "  SIP Transfer mode:      %s\n", transfermode2str(global_allowtransfer));
 	ast_cli(fd, "  Max Call Bitrate:       %d kbps\r\n", default_maxcallbitrate);
+	ast_cli(fd, "  Max Call Bitrate h263p: %d kbps\r\n", default_maxcallbitrate_h263p);
 	ast_cli(fd, "  Auto-Framing:           %s \r\n", global_autoframing ? "Yes" : "No");
 	ast_cli(fd, "\nDefault Settings:\n");
 	ast_cli(fd, "-----------------\n");
@@ -16799,6 +16816,7 @@
 	default_fromdomain[0] = '\0';
 	default_qualify = DEFAULT_QUALIFY;
 	default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
+	default_maxcallbitrate_h263p = DEFAULT_MAX_CALL_BITRATE_H263P;
 	ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
 	ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
 	ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
@@ -17059,6 +17077,10 @@
 			default_maxcallbitrate = atoi(v->value);
 			if (default_maxcallbitrate < 0)
 				default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
+		} else if (!strcasecmp(v->name, "maxcallbitrate_h263p")) {
+			default_maxcallbitrate_h263p = atoi(v->value);
+			if (default_maxcallbitrate_h263p < 0)
+				default_maxcallbitrate_h263p = DEFAULT_MAX_CALL_BITRATE_H263P;
 		} else if (!strcasecmp(v->name, "matchexterniplocally")) {
 			global_matchexterniplocally = ast_true(v->value);
 		}