diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a317879e4baeed8114e6fb88ab9bcb1e3e70cf94..e086711c2cabde260ca0acab2258e137661cd5ca 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11332,25 +11332,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
}
- if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 32000) {
- ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- } else {
- found = TRUE;
- }
- }
- } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 48000) {
- ast_log(LOG_WARNING, "Got Siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- } else {
- found = TRUE;
- }
- }
- } else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
+ if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 64000) {
ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate);
@@ -13009,12 +12991,6 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
} else if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
- } else if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we only expect 32Kbps */
- ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
- } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we only expect 48Kbps */
- ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
} else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we only expect 64Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
diff --git a/res/res_format_attr_siren14.c b/res/res_format_attr_siren14.c
new file mode 100644
index 0000000000000000000000000000000000000000..335b575980ba0a6d3ec4238f5bca6769043e7216
--- /dev/null
+++ b/res/res_format_attr_siren14.c
@@ -0,0 +1,94 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@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 Siren14 format attribute interface
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_REGISTER_FILE()
+
+#include "asterisk/module.h"
+#include "asterisk/format.h"
+
+/* Destroy is a required callback and must exist */
+static void siren14_destroy(struct ast_format *format)
+{
+}
+
+/* Clone is a required callback and must exist */
+static int siren14_clone(const struct ast_format *src, struct ast_format *dst)
+{
+ return 0;
+}
+
+static struct ast_format *siren14_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
+{
+ unsigned int val;
+
+ if (sscanf(attributes, "bitrate=%30u", &val) == 1) {
+ if (val != 48000) {
+ ast_log(LOG_WARNING, "Got siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", val);
+ return NULL;
+ }
+ }
+
+ /* We aren't modifying the format and once passed back it won't be touched, so use what we were given */
+ return ao2_bump((struct ast_format *)format);
+}
+
+static void siren14_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
+{
+ ast_str_append(str, 0, "a=fmtp:%u bitrate=48000\r\n", payload);
+}
+
+static struct ast_format_interface siren14_interface = {
+ .format_destroy = siren14_destroy,
+ .format_clone = siren14_clone,
+ .format_parse_sdp_fmtp = siren14_parse_sdp_fmtp,
+ .format_generate_sdp_fmtp = siren14_generate_sdp_fmtp,
+};
+
+static int load_module(void)
+{
+ if (ast_format_interface_register("siren14", &siren14_interface)) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Siren14 Format Attribute Module",
+ .support_level = AST_MODULE_SUPPORT_CORE,
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);
diff --git a/res/res_format_attr_siren7.c b/res/res_format_attr_siren7.c
new file mode 100644
index 0000000000000000000000000000000000000000..7aef019dabea622826d050f6a95f2ff41812406d
--- /dev/null
+++ b/res/res_format_attr_siren7.c
@@ -0,0 +1,94 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@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 Siren7 format attribute interface
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_REGISTER_FILE()
+
+#include "asterisk/module.h"
+#include "asterisk/format.h"
+
+/* Destroy is a required callback and must exist */
+static void siren7_destroy(struct ast_format *format)
+{
+}
+
+/* Clone is a required callback and must exist */
+static int siren7_clone(const struct ast_format *src, struct ast_format *dst)
+{
+ return 0;
+}
+
+static struct ast_format *siren7_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
+{
+ unsigned int val;
+
+ if (sscanf(attributes, "bitrate=%30u", &val) == 1) {
+ if (val != 32000) {
+ ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", val);
+ return NULL;
+ }
+ }
+
+ /* We aren't modifying the format and once passed back it won't be touched, so use what we were given */
+ return ao2_bump((struct ast_format *)format);
+}
+
+static void siren7_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
+{
+ ast_str_append(str, 0, "a=fmtp:%u bitrate=32000\r\n", payload);
+}
+
+static struct ast_format_interface siren7_interface = {
+ .format_destroy = siren7_destroy,
+ .format_clone = siren7_clone,
+ .format_parse_sdp_fmtp = siren7_parse_sdp_fmtp,
+ .format_generate_sdp_fmtp = siren7_generate_sdp_fmtp,
+};
+
+static int load_module(void)
+{
+ if (ast_format_interface_register("siren7", &siren7_interface)) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Siren7 Format Attribute Module",
+ .support_level = AST_MODULE_SUPPORT_CORE,
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);