diff -urN asterisk-11.1.2/channels/chan_sip.c asterisk-11.1.2-opus-passthrough/channels/chan_sip.c
--- asterisk-11.1.2/channels/chan_sip.c	2013-01-02 20:23:44.000000000 +0100
+++ asterisk-11.1.2-opus-passthrough/channels/chan_sip.c	2013-07-18 11:20:46.519468160 +0200
@@ -7724,8 +7724,24 @@
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */
 		if (p->vrtp && !p->novideo) {
-			transmit_info_with_vidupdate(p);
-			/* ast_rtcp_send_h261fur(p->vrtp); */
+			/* Only use this for WebRTC users */
+			struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
+			struct ast_format vp8;
+			ast_format_set(&vp8, AST_FORMAT_VP8, 0);
+			if(ast_format_cap_iscompatible(fcap, &vp8)) {
+				sip_pvt_lock(p);
+				if (p->vrtp) {
+					/* FIXME Fake RTP write, this will be sent as an RTCP packet */
+					struct ast_frame fr;
+					fr.frametype = AST_FRAME_CONTROL;
+					fr.subclass.integer = AST_CONTROL_VIDUPDATE;
+					res = ast_rtp_instance_write(p->vrtp, &fr);
+				}
+				sip_pvt_unlock(p);
+			} else {
+				transmit_info_with_vidupdate(p);
+				/* ast_rtcp_send_h261fur(p->vrtp); */
+			}
 		} else
 			res = -1;
 		break;
@@ -11041,7 +11057,9 @@
 		/* We have a rtpmap to handle */
 		if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
 			/* Note: should really look at the '#chans' params too */
-			if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
+			if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)
+					/* VP8 */
+					|| !strncasecmp(mimeSubtype, "VP8", 3)) {
 				if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
 					if (debug)
 						ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
@@ -12583,7 +12601,11 @@
 	} else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
 		return;
 	ast_str_append(m_buf, 0, " %d", rtp_code);
-	ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);
+	/* Opus mandates 2 channels in rtpmap */
+	if((int) format->id == AST_FORMAT_OPUS)
+		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d/2\r\n", rtp_code, mime, rate);
+	else
+		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);
 
 	ast_format_sdp_generate(format, rtp_code, a_buf);
 
@@ -12646,6 +12668,10 @@
 
 	ast_str_append(m_buf, 0, " %d", rtp_code);
 	ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, subtype, rate);
+	/* VP8: add RTCP FIR support */
+	if((int) format->id == AST_FORMAT_VP8) {
+		ast_str_append(a_buf, 0, "a=rtcp-fb:* ccm fir\r\n");
+	}
 
 	ast_format_sdp_generate(format, rtp_code, a_buf);
 }
diff -urN asterisk-11.1.2/include/asterisk/format.h asterisk-11.1.2-opus-passthrough/include/asterisk/format.h
--- asterisk-11.1.2/include/asterisk/format.h	2012-07-13 20:41:07.000000000 +0200
+++ asterisk-11.1.2-opus-passthrough/include/asterisk/format.h	2013-07-18 10:35:48.309453118 +0200
@@ -29,6 +29,7 @@
 #include "asterisk/astobj2.h"
 #include "asterisk/silk.h"
 #include "asterisk/celt.h"
+#include "asterisk/opus.h"
 #define AST_FORMAT_ATTR_SIZE 64
 #define AST_FORMAT_INC 100000
 
@@ -101,6 +102,8 @@
 	AST_FORMAT_SLINEAR192       = 27 + AST_FORMAT_TYPE_AUDIO,
 	AST_FORMAT_SPEEX32          = 28 + AST_FORMAT_TYPE_AUDIO,
 	AST_FORMAT_CELT             = 29 + AST_FORMAT_TYPE_AUDIO,
+	/*! Opus */
+	AST_FORMAT_OPUS             = 30 + AST_FORMAT_TYPE_AUDIO,
 
 	/*! H.261 Video */
 	AST_FORMAT_H261             = 1 + AST_FORMAT_TYPE_VIDEO,
@@ -112,6 +115,8 @@
 	AST_FORMAT_H264             = 4 + AST_FORMAT_TYPE_VIDEO,
 	/*! MPEG4 Video */
 	AST_FORMAT_MP4_VIDEO        = 5 + AST_FORMAT_TYPE_VIDEO,
+	/*! VP8 */
+	AST_FORMAT_VP8              = 6 + AST_FORMAT_TYPE_VIDEO,
 
 	/*! JPEG Images */
 	AST_FORMAT_JPEG             = 1 + AST_FORMAT_TYPE_IMAGE,
diff -urN asterisk-11.1.2/include/asterisk/opus.h asterisk-11.1.2-opus-passthrough/include/asterisk/opus.h
--- asterisk-11.1.2/include/asterisk/opus.h	1970-01-01 01:00:00.000000000 +0100
+++ asterisk-11.1.2-opus-passthrough/include/asterisk/opus.h	2013-07-18 11:22:48.810809294 +0200
@@ -0,0 +1,41 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011, Digium, Inc.
+ *
+ * David Vossel <dvossel@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Opus Format Attributes (http://tools.ietf.org/html/draft-ietf-payload-rtp-opus)
+ *
+ * \author Lorenzo Miniero <lorenzo@meetecho.com>
+ */
+#ifndef _AST_FORMAT_OPUS_H_
+#define _AST_FORMAT_OPUS_H_
+
+/*! Opus format attribute key value pairs, all are accessible through ast_format_get_value()*/
+enum opus_attr_keys {
+	OPUS_ATTR_KEY_MAX_BITRATE, /*! value is an int (6000-510000 in spec). */
+	OPUS_ATTR_KEY_MAX_PLAYRATE, /*! value is an int (8000-48000), maximum output rate the receiver can render. */
+	OPUS_ATTR_KEY_MINPTIME, /*! value is an int (3-120 in spec, 10-60 in format.c), decoder's minimum length of time in milliseconds. */
+	OPUS_ATTR_KEY_STEREO, /*! value is an int, 1 prefer receiving stereo, 0 prefer mono. */
+	OPUS_ATTR_KEY_CBR, /*! value is an int, 1 use constant bitrate, 0 use variable bitrate. */
+	OPUS_ATTR_KEY_FEC, /*! value is an int, 1 encode with FEC, 0 do not use FEC. */
+	OPUS_ATTR_KEY_DTX, /*! value is an int, 1 dtx is enabled, 0 dtx not enabled. */
+	OPUS_ATTR_KEY_SPROP_CAPTURE_RATE, /*! value is an int (8000-48000), likely input rate we're going to produce. */
+	OPUS_ATTR_KEY_SPROP_STEREO, /*! value is an int, 1 likely to send stereo, 0 likely to send mono. */
+};
+
+#endif /* _AST_FORMAT_OPUS_H */
diff -urN asterisk-11.1.2/main/channel.c asterisk-11.1.2-opus-passthrough/main/channel.c
--- asterisk-11.1.2/main/channel.c	2013-01-02 20:23:44.000000000 +0100
+++ asterisk-11.1.2-opus-passthrough/main/channel.c	2013-07-03 11:04:06.515699866 +0200
@@ -909,6 +909,8 @@
 		AST_FORMAT_SPEEX32,
 		AST_FORMAT_SPEEX16,
 		AST_FORMAT_SPEEX,
+		/*! Opus */
+		AST_FORMAT_OPUS,
 		/*! SILK is pretty awesome. */
 		AST_FORMAT_SILK,
 		/*! CELT supports crazy high sample rates */
diff -urN asterisk-11.1.2/main/format.c asterisk-11.1.2-opus-passthrough/main/format.c
--- asterisk-11.1.2/main/format.c	2012-10-02 03:27:19.000000000 +0200
+++ asterisk-11.1.2-opus-passthrough/main/format.c	2013-07-03 11:04:12.262613954 +0200
@@ -430,6 +430,9 @@
 	/*! SpeeX Wideband (16kHz) Free Compression */
 	case AST_FORMAT_SPEEX16:
 		return (1ULL << 33);
+	/*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+	case AST_FORMAT_OPUS:
+		return (1ULL << 34);
 	/*! Raw mu-law data (G.711) */
 	case AST_FORMAT_TESTLAW:
 		return (1ULL << 47);
@@ -449,6 +452,9 @@
 	/*! MPEG4 Video */
 	case AST_FORMAT_MP4_VIDEO:
 		return (1ULL << 22);
+	/*! VP8 Video */
+	case AST_FORMAT_VP8:
+		return (1ULL << 23);
 
 	/*! JPEG Images */
 	case AST_FORMAT_JPEG:
@@ -532,6 +538,9 @@
 	/*! SpeeX Wideband (16kHz) Free Compression */
 	case (1ULL << 33):
 		return ast_format_set(dst, AST_FORMAT_SPEEX16, 0);
+	/*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+	case (1ULL << 34):
+		return ast_format_set(dst, AST_FORMAT_OPUS, 0);
 	/*! Raw mu-law data (G.711) */
 	case (1ULL << 47):
 		return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);
@@ -551,6 +560,9 @@
 	/*! MPEG4 Video */
 	case (1ULL << 22):
 		return ast_format_set(dst, AST_FORMAT_MP4_VIDEO, 0);
+	/*! VP8 Video */
+	case (1ULL << 23):
+		return ast_format_set(dst, AST_FORMAT_VP8, 0);
 
 	/*! JPEG Images */
 	case (1ULL << 16):
@@ -782,6 +794,9 @@
 			return samplerate;
 		}
 	}
+	/* Opus */
+	case AST_FORMAT_OPUS:
+		return 48000;
 	default:
 		return 8000;
 	}
@@ -1072,6 +1087,10 @@
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), "slin48", 48000, "16 bit Signed Linear PCM (48kHz)", 960, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (48kHz) */
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR96, 0), "slin96", 96000, "16 bit Signed Linear PCM (96kHz)", 1920, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (96kHz) */
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR192, 0), "slin192", 192000, "16 bit Signed Linear PCM (192kHz)", 3840, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (192kHz) */
+	/* Opus (FIXME: real min is 3/5/10, real max is 120...) */
+	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), "opus", 48000, "Opus Codec", 10, 20, 60, 20, 20, 0, 0);   /*!< codec_opus.c */
+	/* VP8 (passthrough) */
+	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), "vp8", 0, "VP8 Video", 0, 0, 0, 0 ,0 ,0, 0);         /*!< Passthrough support, see format_h263.c */
 
 	return 0;
 }
diff -urN asterisk-11.1.2/main/frame.c asterisk-11.1.2-opus-passthrough/main/frame.c
--- asterisk-11.1.2/main/frame.c	2012-07-24 18:54:26.000000000 +0200
+++ asterisk-11.1.2-opus-passthrough/main/frame.c	2013-07-18 11:56:16.675089967 +0200
@@ -1083,6 +1083,10 @@
 		/* TODO The assumes 20ms delivery right now, which is incorrect */
 		samples = ast_format_rate(&f->subclass.format) / 50;
 		break;
+	case AST_FORMAT_OPUS:
+		/* TODO The assumes 20ms delivery right now, which is incorrect */
+		samples = 960;
+		break;
 	default:
 		ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
 	}
diff -urN asterisk-11.1.2/main/rtp_engine.c asterisk-11.1.2-opus-passthrough/main/rtp_engine.c
--- asterisk-11.1.2/main/rtp_engine.c	2012-09-20 20:18:47.000000000 +0200
+++ asterisk-11.1.2-opus-passthrough/main/rtp_engine.c	2013-07-03 11:04:21.529480072 +0200
@@ -2268,6 +2268,9 @@
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+	/* Opus and VP8 */
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0,  "audio", "opus", 48000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0,  "video", "VP8", 90000);
 
 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
@@ -2309,6 +2312,9 @@
 	add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
 	add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
 	add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
+	/* Opus and VP8 */
+	add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
+	add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
 
 	return 0;
 }
diff -urN asterisk-11.1.2/res/res_format_attr_opus.c asterisk-11.1.2-opus-passthrough/res/res_format_attr_opus.c
--- asterisk-11.1.2/res/res_format_attr_opus.c	1970-01-01 01:00:00.000000000 +0100
+++ asterisk-11.1.2-opus-passthrough/res/res_format_attr_opus.c	2013-07-18 15:14:41.701160708 +0200
@@ -0,0 +1,322 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2011, Digium, Inc.
+ *
+ * David Vossel <dvossel@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Opus format attribute interface
+ *
+ * \author Lorenzo Miniero <lorenzo@meetecho.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 370055 $")
+
+#include "asterisk/module.h"
+#include "asterisk/format.h"
+
+/*!
+ * \brief Opus attribute structure.
+ *
+ * \note http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00.
+ */
+struct opus_attr {
+	unsigned int maxbitrate;	/* Default 64-128 kb/s for FB stereo music */
+	unsigned int maxplayrate	/* Default 48000 */;
+	unsigned int minptime;	/* Default 3, but it's 10 in format.c */
+	unsigned int stereo;	/* Default 0 */
+	unsigned int cbr;	/* Default 0 */
+	unsigned int fec;	/* Default 0 */
+	unsigned int dtx;	/* Default 0 */
+	unsigned int spropmaxcapturerate;	/* Default 48000 */
+	unsigned int spropstereo;	/* Default 0 */
+};
+
+static int opus_sdp_parse(struct ast_format_attr *format_attr, const char *attributes)
+{
+	struct opus_attr *attr = (struct opus_attr *) format_attr;
+	unsigned int val;
+
+	if (strstr(attributes, "maxplaybackrate") && sscanf(strstr(attributes, "maxplaybackrate"), "maxplaybackrate=%30u", &val) == 1) {
+		attr->maxplayrate = val;
+	}
+	if (strstr(attributes, "sprop-maxcapturerate") && sscanf(strstr(attributes, "sprop-maxcapturerate"), "sprop-maxcapturerate=%30u", &val) == 1) {
+		attr->spropmaxcapturerate = val;
+	}
+	if (strstr(attributes, "minptime") && sscanf(strstr(attributes, "minptime"), "minptime=%30u", &val) == 1) {
+		attr->minptime = val;
+	}
+	if (strstr(attributes, "maxaveragebitrate") && sscanf(strstr(attributes, "maxaveragebitrate"), "maxaveragebitrate=%30u", &val) == 1) {
+		attr->maxbitrate = val;
+	}
+	if (strstr(attributes, " stereo") && sscanf(strstr(attributes, " stereo"), " stereo=%30u", &val) == 1) {
+		attr->stereo = val;
+	}
+	if (strstr(attributes, ";stereo") && sscanf(strstr(attributes, ";stereo"), ";stereo=%30u", &val) == 1) {
+		attr->stereo = val;
+	}
+	if (strstr(attributes, "sprop-stereo") && sscanf(strstr(attributes, "sprop-stereo"), "sprop-stereo=%30u", &val) == 1) {
+		attr->spropstereo = val;
+	}
+	if (strstr(attributes, "cbr") && sscanf(strstr(attributes, "cbr"), "cbr=%30u", &val) == 1) {
+		attr->cbr = val;
+	}
+	if (strstr(attributes, "useinbandfec") && sscanf(strstr(attributes, "useinbandfec"), "useinbandfec=%30u", &val) == 1) {
+		attr->fec = val;
+	}
+	if (strstr(attributes, "usedtx") && sscanf(strstr(attributes, "usedtx"), "usedtx=%30u", &val) == 1) {
+		attr->dtx = val;
+	}
+
+	return 0;
+}
+
+static void opus_sdp_generate(const struct ast_format_attr *format_attr, unsigned int payload, struct ast_str **str)
+{
+	struct opus_attr *attr = (struct opus_attr *) format_attr;
+
+	/* FIXME should we only generate attributes that were explicitly set? */
+	ast_str_append(str, 0,
+				"a=fmtp:%d "
+					"maxplaybackrate=%d;"
+					"sprop-maxcapturerate=%d;"
+					"minptime=%d;"
+					"maxaveragebitrate=%d;"
+					"stereo=%d;"
+					"sprop-stereo=%d;"
+					"cbr=%d;"
+					"useinbandfec=%d;"
+					"usedtx=%d\r\n",
+			payload,
+			attr->maxplayrate ? attr->maxplayrate : 48000,	/* maxplaybackrate */
+			attr->spropmaxcapturerate ? attr->spropmaxcapturerate : 48000,	/* sprop-maxcapturerate */
+			attr->minptime > 10 ? attr->minptime : 10,	/* minptime */
+			attr->maxbitrate ? attr->maxbitrate : 20000,	/* maxaveragebitrate */
+			attr->stereo ? 1 : 0,		/* stereo */
+			attr->spropstereo ? 1 : 0,		/* sprop-stereo */
+			attr->cbr ? 1 : 0,		/* cbr */
+			attr->fec ? 1 : 0,		/* useinbandfec */
+			attr->dtx ? 1 : 0		/* usedtx */
+	);
+	/* FIXME what about maxptime? that's not an Opus specific attribute */
+	ast_str_append(str, 0, "a=maxptime:%d\r\n", 60);	/* As defined in format.c */
+}
+
+static int opus_get_val(const struct ast_format_attr *fattr, int key, void *result)
+{
+	const struct opus_attr *attr = (struct opus_attr *) fattr;
+	int *val = result;
+
+	switch (key) {
+	case OPUS_ATTR_KEY_MAX_BITRATE:
+		*val = attr->maxbitrate;
+		break;
+	case OPUS_ATTR_KEY_MAX_PLAYRATE:
+		*val = attr->maxplayrate;
+		break;
+	case OPUS_ATTR_KEY_MINPTIME:
+		*val = attr->minptime;
+		break;
+	case OPUS_ATTR_KEY_STEREO:
+		*val = attr->stereo;
+		break;
+	case OPUS_ATTR_KEY_CBR:
+		*val = attr->cbr;
+		break;
+	case OPUS_ATTR_KEY_FEC:
+		*val = attr->fec;
+		break;
+	case OPUS_ATTR_KEY_DTX:
+		*val = attr->dtx;
+		break;
+	case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+		*val = attr->spropmaxcapturerate;
+		break;
+	case OPUS_ATTR_KEY_SPROP_STEREO:
+		*val = attr->spropstereo;
+		break;
+	default:
+		ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+		return -1;
+	}
+	return 0;
+}
+
+static int opus_isset(const struct ast_format_attr *fattr, va_list ap)
+{
+	enum opus_attr_keys key;
+	const struct opus_attr *attr = (struct opus_attr *) fattr;
+
+	for (key = va_arg(ap, int);
+		key != AST_FORMAT_ATTR_END;
+		key = va_arg(ap, int))
+	{
+		switch (key) {
+		case OPUS_ATTR_KEY_MAX_BITRATE:
+			if (attr->maxbitrate != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_MAX_PLAYRATE:
+			if (attr->maxplayrate != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_MINPTIME:
+			if (attr->minptime != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_STEREO:
+			if (attr->stereo != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_CBR:
+			if (attr->cbr != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_FEC:
+			if (attr->fec != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_DTX:
+			if (attr->dtx != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+			if (attr->spropmaxcapturerate != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		case OPUS_ATTR_KEY_SPROP_STEREO:
+			if (attr->spropstereo != (va_arg(ap, int))) {
+				return -1;
+			}
+			break;
+		default:
+			ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+			return -1;
+		}
+	}
+	return 0;
+}
+static int opus_getjoint(const struct ast_format_attr *fattr1, const struct ast_format_attr *fattr2, struct ast_format_attr *result)
+{
+	struct opus_attr *attr1 = (struct opus_attr *) fattr1;
+	struct opus_attr *attr2 = (struct opus_attr *) fattr2;
+	struct opus_attr *attr_res = (struct opus_attr *) result;
+	int joint = 0;
+
+	/* Only do dtx if both sides want it. DTX is a trade off between
+	 * computational complexity and bandwidth. */
+	attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
+
+	/* Only do FEC if both sides want it.  If a peer specifically requests not
+	 * to receive with FEC, it may be a waste of bandwidth. */
+	attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
+
+	/* Only do stereo if both sides want it.  If a peer specifically requests not
+	 * to receive stereo signals, it may be a waste of bandwidth. */
+	attr_res->stereo = attr1->stereo && attr2->stereo ? 1 : 0;
+	
+	/* FIXME: do we need to join other attributes as well, e.g., minptime, cbr, etc.? */
+
+	return joint;
+}
+
+static void opus_set(struct ast_format_attr *fattr, va_list ap)
+{
+	enum opus_attr_keys key;
+	struct opus_attr *attr = (struct opus_attr *) fattr;
+
+	for (key = va_arg(ap, int);
+		key != AST_FORMAT_ATTR_END;
+		key = va_arg(ap, int))
+	{
+		switch (key) {
+		case OPUS_ATTR_KEY_MAX_BITRATE:
+			attr->maxbitrate = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_MAX_PLAYRATE:
+			attr->maxplayrate = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_MINPTIME:
+			attr->minptime = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_STEREO:
+			attr->stereo = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_CBR:
+			attr->cbr = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_FEC:
+			attr->fec = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_DTX:
+			attr->dtx = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_SPROP_CAPTURE_RATE:
+			attr->spropmaxcapturerate = (va_arg(ap, int));
+			break;
+		case OPUS_ATTR_KEY_SPROP_STEREO:
+			attr->spropstereo = (va_arg(ap, int));
+			break;
+		default:
+			ast_log(LOG_WARNING, "unknown attribute type %d\n", key);
+		}
+	}
+}
+
+static struct ast_format_attr_interface opus_interface = {
+	.id = AST_FORMAT_OPUS,
+	.format_attr_get_joint = opus_getjoint,
+	.format_attr_set = opus_set,
+	.format_attr_isset = opus_isset,
+	.format_attr_get_val = opus_get_val,
+	.format_attr_sdp_parse = opus_sdp_parse,
+	.format_attr_sdp_generate = opus_sdp_generate,
+};
+
+static int load_module(void)
+{
+	if (ast_format_attr_reg_interface(&opus_interface)) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_format_attr_unreg_interface(&opus_interface);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Opus Format Attribute Module",
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);
diff -urN asterisk-11.1.2/res/res_rtp_asterisk.c asterisk-11.1.2-opus-passthrough/res/res_rtp_asterisk.c
--- asterisk-11.1.2/res/res_rtp_asterisk.c	2012-10-11 18:04:19.000000000 +0200
+++ asterisk-11.1.2-opus-passthrough/res/res_rtp_asterisk.c	2013-07-18 12:06:29.406813101 +0200
@@ -91,6 +91,8 @@
 #define RTCP_PT_SDES    202
 #define RTCP_PT_BYE     203
 #define RTCP_PT_APP     204
+/* VP8: RTCP Feedback */
+#define RTCP_PT_PSFB    206
 
 #define RTP_MTU		1200
 
@@ -341,6 +343,9 @@
 	double normdevrtt;
 	double stdevrtt;
 	unsigned int rtt_count;
+
+	/* VP8: sequence number for the RTCP FIR FCI */
+	int firseq;
 };
 
 struct rtp_red {
@@ -2599,6 +2604,39 @@
 		return 0;
 	}
 
+	/* VP8: is this a request to send a RTCP FIR? */
+	if(frame->frametype == AST_FRAME_CONTROL && frame->subclass.integer == AST_CONTROL_VIDUPDATE) {
+		struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+		if (!rtp || !rtp->rtcp)
+			return 0;
+		unsigned int *rtcpheader;
+		char bdata[1024];
+		if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
+			/*
+			 * RTCP was stopped.
+			 */
+			return 0;
+		}
+		/* Prepare RTCP FIR (PT=206, FMT=4) */
+		rtp->rtcp->firseq++;
+		if(rtp->rtcp->firseq == 256)
+			rtp->rtcp->firseq = 0;
+		int len = 20;
+		int ice;
+		rtcpheader = (unsigned int *)bdata;
+		rtcpheader[0] = htonl((2 << 30) | (4 << 24) | (RTCP_PT_PSFB << 16) | ((len/4)-1));
+		rtcpheader[1] = htonl(rtp->ssrc);
+		rtcpheader[2] = htonl(rtp->themssrc);
+		rtcpheader[3] = htonl(rtp->themssrc);	/* FCI: SSRC */
+		rtcpheader[4] = htonl(rtp->rtcp->firseq << 24);			/* FCI: Sequence number */
+		int res = rtcp_sendto(instance, (unsigned int *)rtcpheader, len, 0, &rtp->rtcp->them, &ice);
+		if (res < 0) {
+			ast_log(LOG_ERROR, "RTCP FIR transmission error: %s\n",strerror(errno));
+			return 0;
+		}
+		return 0;
+	}
+
 	/* If there is no data length we can't very well send the packet */
 	if (!frame->datalen) {
 		ast_debug(1, "Received frame with no data for RTP instance '%p' so dropping frame\n", instance);
@@ -2650,6 +2688,8 @@
 		case AST_FORMAT_SIREN7:
 		case AST_FORMAT_SIREN14:
 		case AST_FORMAT_G719:
+		/* Opus */
+		case AST_FORMAT_OPUS:
 			/* these are all frame-based codecs and cannot be safely run through
 			   a smoother */
 			break;
@@ -3308,6 +3348,8 @@
 			}
 			break;
 		case RTCP_PT_FUR:
+		/* Handle RTCP FIR as FUR */
+		case RTCP_PT_PSFB:
 			if (rtcp_debug_test_addr(&addr))
 				ast_verbose("Received an RTCP Fast Update Request\n");
 			rtp->f.frametype = AST_FRAME_CONTROL;