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- make sure all RTP ports are unsigned
- prevents totally bogus SDP offers from being made due to conversion kabooms (m=audio 4294935774 RTP/AVP 3) (upstream r8160)
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@@ -170,7 +170,7 @@ void TranEntry::setDialog(SIP::SipDialog *dialog) { mDialog = dialog; dialog->se
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void TranEntry::txFrame(GSM::AudioFrame* frame, unsigned numFlushed) { getDialog()->txFrame(frame,numFlushed); }
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GSM::AudioFrame *TranEntry::rxFrame() { return getDialog()->rxFrame(); } // Crashes if rtp not established.
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short TranEntry::getRTPPort() const
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unsigned TranEntry::getRTPPort() const
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{
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if (SipDialog *dialog = getDialog()) { return dialog->RTPPort(); }
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return 0;
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@@ -448,7 +448,7 @@ public:
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//@{
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//unused SIP::SipState getSipState() const;
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short getRTPPort() const;
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unsigned getRTPPort() const;
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// Obviously, these are only for TransactionEntries for voice calls.
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@@ -251,7 +251,7 @@ class SipRtp {
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public:
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/**@name RTP state and parameters. */
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//@{
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short mRTPPort;
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unsigned mRTPPort;
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//short mRTPRemPort;
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//string mRTPRemIP;
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Control::CodecSet mCodec;
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@@ -275,7 +275,7 @@ class SipRtp {
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RtpSession * RTPSession() const { return mSession; }
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/** Return the RTP Port being used. */
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short RTPPort() const { return mRTPPort; }
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unsigned RTPPort() const { return mRTPPort; }
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bool txDtmf();
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