From: Alexander Traud <pabstraud@compuserve.com>
Date: Fri, 6 Nov 2015 12:57:15 +0000 (+0100)
Subject: res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.
X-Git-Tag: 13.7.0-rc1~79^2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=refs%2Fchanges%2F75%2F1575%2F1;p=thirdparty%2Fasterisk.git

res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.

In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual
amount of channels is negotiated in-band. Therefore now, the Opus codec and its
attribute rtpmap are registered with two channels.

ASTERISK-24779 #close
Reported by: PowerPBX
Tested by: Alexander Traud
patches:
  asterisk-24779.patch submitted by Sean Bright (license #5060)

Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b
---

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index df2820479e..d031285f08 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -387,8 +387,11 @@ static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, p
 	rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
 	rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
 	pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
-	rtpmap.param.slen = 0;
-	rtpmap.param.ptr = NULL;
+	if (!pj_stricmp2(&rtpmap.enc_name, "opus")) {
+		pj_cstr(&rtpmap.param, "2");
+	} else {
+		pj_cstr(&rtpmap.param, NULL);
+	}
 
 	pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);