- 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)
This commit is contained in:
Michael Iedema
2014-04-09 15:23:55 +02:00
parent e62971ef41
commit 3b79030258
3 changed files with 4 additions and 4 deletions

View File

@@ -170,7 +170,7 @@ void TranEntry::setDialog(SIP::SipDialog *dialog) { mDialog = dialog; dialog->se
void TranEntry::txFrame(GSM::AudioFrame* frame, unsigned numFlushed) { getDialog()->txFrame(frame,numFlushed); }
GSM::AudioFrame *TranEntry::rxFrame() { return getDialog()->rxFrame(); } // Crashes if rtp not established.
short TranEntry::getRTPPort() const
unsigned TranEntry::getRTPPort() const
{
if (SipDialog *dialog = getDialog()) { return dialog->RTPPort(); }
return 0;

View File

@@ -448,7 +448,7 @@ public:
//@{
//unused SIP::SipState getSipState() const;
short getRTPPort() const;
unsigned getRTPPort() const;
// Obviously, these are only for TransactionEntries for voice calls.

View File

@@ -251,7 +251,7 @@ class SipRtp {
public:
/**@name RTP state and parameters. */
//@{
short mRTPPort;
unsigned mRTPPort;
//short mRTPRemPort;
//string mRTPRemIP;
Control::CodecSet mCodec;
@@ -275,7 +275,7 @@ class SipRtp {
RtpSession * RTPSession() const { return mSession; }
/** Return the RTP Port being used. */
short RTPPort() const { return mRTPPort; }
unsigned RTPPort() const { return mRTPPort; }
bool txDtmf();