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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 19f8aa308a801fe3c7072a0027f27a62b6302c7c..d44bf8a83e3d6fd73d46f35a4f69d6c669c9d7b3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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);
        }