diff options
-rw-r--r-- | channels/chan_sip.c | 5 | ||||
-rw-r--r-- | res/res_format_attr_g729.c | 76 |
2 files changed, 77 insertions, 4 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index f57d4670b..6250731bd 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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) { 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, +); |