From: Mark Michelson Date: Fri, 29 Jul 2016 18:13:55 +0000 (-0500) Subject: Remove SILK payload mappings from Asterisk core. X-Git-Tag: 13.12.0-rc1~156^2 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=f6821fbaec3fed7bbc1c814de3a4824cc926a90d;p=thirdparty%2Fasterisk.git 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 (cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817) --- diff --git a/main/rtp_engine.c b/main/rtp_engine.c index 50398a5c6f..db733acf8a 100644 --- a/main/rtp_engine.c +++ b/main/rtp_engine.c @@ -2200,11 +2200,6 @@ 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); @@ -2250,11 +2245,6 @@ 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; }