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Even if configured start RTP when generating 200 OK only if INVITE had SDP. 

git-svn-id: http://voip.null.ro/svn/yate@6036 acf43c95-373e-0410-b603-e72c3f656dc1
This commit is contained in:
paulc 2015-09-16 08:11:49 +00:00
parent 66379a94ca
commit 48c5f91ffb
1 changed files with 4 additions and 2 deletions
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6
modules/ysipchan.cpp
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@ -7387,8 +7387,10 @@ bool YateSIPConnection::msgAnswered(Message& msg)
MimeSdpBody* sdp = createPasstroughSDP(msg);
if (!sdp) {
m_rtpForward = false;
bool startNow = msg.getBoolValue(YSTRING("rtp_start"),s_start_rtp);
if (startNow && !m_rtpMedia) {
bool startNow = false;
if (m_rtpMedia)
startNow = msg.getBoolValue(YSTRING("rtp_start"),s_start_rtp);
else {
// early RTP start but media list yet unknown - build best guess
String fmts;
plugin.parser().getAudioFormats(fmts);