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authorAlexander Traud <pabstraud@compuserve.com>2016-06-08 09:11:40 +0200
committerAlexander Traud <pabstraud@compuserve.com>2016-06-08 09:13:01 +0200
commit784c18128b4b96c9993e7cfff93a70b75096762c (patch)
tree383c3f6291d2a816a2ce2a9b1c04c842925d1307
parent33787459c3400ccf8c52361e0c15a8b450d2285d (diff)
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
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 19f8aa308..d44bf8a83 100644
--- 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);
}