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

@@ -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;
 }