From 1cd79d6ee5e9fa53668a15e2fc5be2b7e6820b1f Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Fri, 29 Jul 2016 13:13:55 -0500 Subject: [PATCH] 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 --- main/rtp_engine.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/main/rtp_engine.c b/main/rtp_engine.c index 66b80e5557..c2b1c8c2b7 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; } -- 2.47.2