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 50398a5c6f4bcab081803280a02fde6e69cc5d97..db733acf8a8a1cd6170dcbeecf365b29f6a16974 100644 (file)
--- 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;
 }