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	Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs"
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		| @@ -12982,10 +12982,7 @@ 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) { | ||||
|   | ||||
							
								
								
									
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								res/res_format_attr_g729.c
									
									
									
									
									
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							| @@ -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, | ||||
| ); | ||||
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