Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs" into 13

2 files changed, 77 insertions, 4 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f58c19e6d..c0aae46c5 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12981,10 +12981,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
 
 	framing = ast_format_cap_get_format_framing(p->caps, format);
 
-	if (ast_format_cmp(format, ast_format_g729) == AST_FORMAT_CMP_EQUAL) {
-		/* Indicate that we don't support VAD (G.729 annex B) */
-		ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
-	} else if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) {
+	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_g719) == AST_FORMAT_CMP_EQUAL) {
diff --git a/res/res_format_attr_g729.c b/res/res_format_attr_g729.c
new file mode 100644
index 000000000..5ba4920d9
--- /dev/null
+++ b/res/res_format_attr_g729.c
@@ -0,0 +1,76 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Jason Parker <jparker@sangoma.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.
+ */
+
+/*** 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 g729_destroy(struct ast_format *format)
+{
+}
+
+/* Clone is a required callback and must exist */
+static int g729_clone(const struct ast_format *src, struct ast_format *dst)
+{
+	return 0;
+}
+
+static void g729_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
+{
+	/*
+	 * According to the rfc the joint annexb format parameter should be set to 'yes'
+	 * or 'no' based on the answerer (rfc7261 - 3.3). However, Asterisk being a B2BUA
+	 * makes things tricky. So for now Asterisk will set annexb=no.
+	 */
+	ast_str_append(str, 0, "a=fmtp:%u annexb=no\r\n", payload);
+}
+
+static struct ast_format_interface g729_interface = {
+	.format_destroy = g729_destroy,
+	.format_clone = g729_clone,
+	.format_generate_sdp_fmtp = g729_generate_sdp_fmtp,
+};
+
+static int load_module(void)
+{
+	if (ast_format_interface_register("g729", &g729_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, "G.729 Format Attribute Module",
+	.support_level = AST_MODULE_SUPPORT_CORE,
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
+);