mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-23 04:58:48 +00:00
chan_sip: No rtpmap for static RTP payload IDs in SDP.
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compactheaders=yes via the file sip.conf. ASTERISK-25578 #close Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044
This commit is contained in:
@@ -12996,7 +12996,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
|
||||
/* Opus mandates 2 channels in rtpmap */
|
||||
if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) {
|
||||
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
|
||||
} else {
|
||||
} else if ((35 <= rtp_code) || !(sip_cfg.compactheaders)) {
|
||||
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
|
||||
}
|
||||
|
||||
|
Reference in New Issue
Block a user