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Only bitmaskify the RTP payload structure for video if an RTP structure exists for it... otherwise the default values will cause codec combination madness
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -4845,7 +4845,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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/* Now gather all of the codecs that we are asked for: */
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ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
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ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
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if (p->vrtp)
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ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
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newjointcapability = p->capability & (peercapability | vpeercapability);
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newpeercapability = (peercapability | vpeercapability);
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