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authorMark Michelson <mmichelson@digium.com>2016-07-29 13:13:55 -0500
committerMark Michelson <mmichelson@digium.com>2016-07-29 13:18:06 -0500
commit1cd79d6ee5e9fa53668a15e2fc5be2b7e6820b1f (patch)
tree6d6839252a6304c611f11e1cb4e93e286937240b
parentd462cb73a8f3d2802bd5261cc2d218a0890d4f1e (diff)
Remove SILK payload mappings from Asterisk core.
SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612
-rw-r--r--main/rtp_engine.c10
1 files changed, 0 insertions, 10 deletions
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 66b80e555..c2b1c8c2b 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -2692,11 +2692,6 @@ int ast_rtp_engine_init(void)
/* Opus and VP8 */
set_next_mime_type(ast_format_opus, 0, "audio", "opus", 48000);
set_next_mime_type(ast_format_vp8, 0, "video", "VP8", 90000);
- /* DA SILK */
- set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
- set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
- set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
- set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);
/* Define the static rtp payload mappings */
add_static_payload(0, ast_format_ulaw, 0);
@@ -2750,11 +2745,6 @@ int ast_rtp_engine_init(void)
add_static_payload(100, ast_format_vp8, 0);
add_static_payload(107, ast_format_opus, 0);
- add_static_payload(108, ast_format_silk8, 0);
- add_static_payload(109, ast_format_silk12, 0);
- add_static_payload(113, ast_format_silk16, 0);
- add_static_payload(114, ast_format_silk24, 0);
-
return 0;
}