From: Alexander Traud Date: Wed, 8 Jun 2016 07:11:40 +0000 (+0200) Subject: chan_sip: No rtpmap for static RTP payload IDs in SDP. X-Git-Tag: 14.0.0-beta1~123^2 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=784c18128b4b96c9993e7cfff93a70b75096762c;p=thirdparty%2Fasterisk.git 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 19f8aa308a..d44bf8a83e 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); }