[Asterisk-code-review] res pjsip sdp rtp: Enable Opus to be negotiated via SIP/SDP. (asterisk[13])

[Alexander Traud]

Alexander Traud has uploaded a new change for review.

  https://gerrit.asterisk.org/1575

Change subject: res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.
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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
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 5 insertions(+), 2 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/75/1575/1

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index df28204..d031285 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -387,8 +387,11 @@
 	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);
 

-- 
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Gerrit-MessageType: newchange
Gerrit-Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>