/usr/include/openh323/rtp.h is in libh323plus-dev 1.21.0~dfsg-1build1.
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* rtp.h
*
* RTP protocol handler
*
* Open H323 Library
*
* Copyright (c) 1998-2000 Equivalence Pty. Ltd.
*
* The contents of this file are subject to the Mozilla Public License
* Version 1.0 (the "License"); you may not use this file except in
* compliance with the License. You may obtain a copy of the License at
* http://www.mozilla.org/MPL/
*
* Software distributed under the License is distributed on an "AS IS"
* basis, WITHOUT WARRANTY OF ANY KIND, either express or implied. See
* the License for the specific language governing rights and limitations
* under the License.
*
* The Original Code is Open H323 Library.
*
* The Initial Developer of the Original Code is Equivalence Pty. Ltd.
*
* Portions of this code were written with the assisance of funding from
* Vovida Networks, Inc. http://www.vovida.com.
*
* Contributor(s): ______________________________________.
*
* $Log: rtp.h,v $
* Revision 1.5 2008/05/23 11:20:01 willamowius
* switch BOOL to PBoolean to be able to compile with Ptlib 2.2.x
*
* Revision 1.4 2008/02/06 02:52:59 shorne
* Added support for Standards based NAT Traversal
*
* Revision 1.3 2008/01/22 01:10:32 shorne
* Fix H.224 opening unidirectional channel
*
* Revision 1.2 2007/10/16 17:03:53 shorne
* Qos capability negotiation
*
* Revision 1.1 2007/08/06 20:50:50 shorne
* First commit of h323plus
*
* Revision 1.60.2.4 2007/07/23 21:47:10 shorne
* Added QoS GK Reporting
*
* Revision 1.60.2.3 2007/03/24 23:39:42 shorne
* More H.239 work
*
* Revision 1.60.2.2 2007/02/17 15:26:20 shorne
* Support static memory allocation of RTP_DataFrame for Plugin Video Support
*
* Revision 1.60.2.1 2006/12/23 19:08:02 shorne
* Plugin video codecs & sundry
*
* Revision 1.60 2006/01/26 03:35:03 shorne
* Added more NAT traversal support
*
* Revision 1.59 2006/01/18 07:46:08 csoutheren
* Initial version of RTP aggregation (disabled by default)
*
* Revision 1.58 2005/11/30 13:05:01 csoutheren
* Changed tags for Doxygen
*
* Revision 1.57 2005/11/21 21:06:23 shorne
* Added GQoS switches
*
* Revision 1.56 2005/08/27 02:06:00 csoutheren
* Keep track of time first RTP packet is received
*
* Revision 1.55 2005/01/03 14:03:20 csoutheren
* Added new configure options and ability to disable/enable modules
*
* Revision 1.54 2005/01/03 06:25:52 csoutheren
* Added extensive support for disabling code modules at compile time
*
* Revision 1.53 2004/04/22 12:38:02 rjongbloed
* Removed the RTP QoS class if there is no QoS support in build,
* thanks Nick Hoath, ISDN Communications Ltd.
*
* Revision 1.52 2004/04/07 05:31:42 csoutheren
* Added ability to receive calls from endpoints behind NAT firewalls
*
* Revision 1.51 2003/10/27 06:03:39 csoutheren
* Added support for QoS
* Thanks to Henry Harrison of AliceStreet
*
* Revision 1.50 2003/10/09 09:47:45 csoutheren
* Fixed problem with re-opening RTP half-channels under unusual
* circumstances. Thanks to Damien Sandras
*
* Revision 1.49 2003/05/14 13:46:39 rjongbloed
* Removed hack of using special payload type for H.263 for a method which
* would be less prone to failure in the future.
*
* Revision 1.48 2003/05/02 04:57:43 robertj
* Added header extension support to RTP data frame class.
*
* Revision 1.47 2003/05/02 04:21:53 craigs
* Added hacked OpalH263 codec
*
* Revision 1.46 2003/02/07 00:30:41 robertj
* Changes for bizarre usage of RTP code outside of scope of H.323 specs.
*
* Revision 1.45 2003/02/05 06:32:08 robertj
* Fixed non-stun symmetric NAT support recently broken.
*
* Revision 1.44 2003/02/04 07:06:41 robertj
* Added STUN support.
*
* Revision 1.43 2002/11/19 01:48:15 robertj
* Allowed get/set of canonical anme and tool name.
*
* Revision 1.42 2002/11/12 22:10:48 robertj
* Updated documentation.
*
* Revision 1.41 2002/10/31 00:33:29 robertj
* Enhanced jitter buffer system so operates dynamically between minimum and
* maximum values. Altered API to assure app writers note the change!
*
* Revision 1.40 2002/09/26 04:01:58 robertj
* Fixed calculation of fraction of packets lost in RR, thanks Awais Ali
*
* Revision 1.39 2002/09/16 01:14:15 robertj
* Added #define so can select if #pragma interface/implementation is used on
* platform basis (eg MacOS) rather than compiler, thanks Robert Monaghan.
*
* Revision 1.38 2002/09/03 05:47:02 robertj
* Normalised the multi-include header prevention ifdef/define symbol.
* Added copy constructor/operator for session manager.
*
* Revision 1.37 2002/08/05 10:03:47 robertj
* Cosmetic changes to normalise the usage of pragma interface/implementation.
*
* Revision 1.36 2002/05/28 02:37:37 robertj
* Fixed reading data out of RTCP compound statements.
*
* Revision 1.35 2002/05/02 05:58:24 robertj
* Changed the mechanism for sending RTCP reports so that they will continue
* to be sent regardless of if there is any actual data traffic.
* Added support for compound RTCP statements for sender and receiver reports.
*
* Revision 1.34 2002/02/09 02:33:37 robertj
* Improved payload type docuemntation and added Cisco CN.
*
* Revision 1.33 2002/01/22 07:08:26 robertj
* Added IllegalPayloadType enum as need marker for none set
* and MaxPayloadType is a legal value.
*
* Revision 1.32 2001/11/09 05:39:54 craigs
* Added initial T.38 support thanks to Adam Lazur
*
* Revision 1.31 2001/09/11 00:21:21 robertj
* Fixed missing stack sizes in endpoint for cleaner thread and jitter thread.
*
* Revision 1.30 2001/07/06 06:32:22 robertj
* Added flag and checks for RTP data having specific SSRC.
* Changed transmitter IP address check so is based on first received
* PDU instead of expecting it to come from the host we are sending to.
*
* Revision 1.29 2001/06/04 11:37:48 robertj
* Added thread safe enumeration functions of RTP sessions.
* Added member access functions to UDP based RTP sessions.
*
* Revision 1.28 2001/04/24 06:15:50 robertj
* Added work around for strange Cisco bug which suddenly starts sending
* RTP packets beginning at a difference sequence number base.
*
* Revision 1.27 2001/04/02 23:58:23 robertj
* Added jitter calculation to RTP session.
* Added trace of statistics.
*
* Revision 1.26 2001/02/09 05:16:24 robertj
* Added #pragma interface for GNU C++.
*
* Revision 1.25 2000/12/18 08:58:30 craigs
* Added ability set ports
*
* Revision 1.24 2000/09/25 01:44:31 robertj
* Fixed possible race condition on shutdown of RTP session with jitter buffer.
*
* Revision 1.23 2000/09/21 02:06:06 craigs
* Added handling for endpoints that return conformant, but useless, RTP address
* and port numbers
*
* Revision 1.22 2000/05/23 12:57:28 robertj
* Added ability to change IP Type Of Service code from applications.
*
* Revision 1.21 2000/05/18 11:53:34 robertj
* Changes to support doc++ documentation generation.
*
* Revision 1.20 2000/05/04 11:49:21 robertj
* Added Packets Too Late statistics, requiring major rearrangement of jitter buffer code.
*
* Revision 1.19 2000/05/02 04:32:25 robertj
* Fixed copyright notice comment.
*
* Revision 1.18 2000/05/01 01:01:24 robertj
* Added flag for what to do with out of orer packets (use if jitter, don't if not).
*
* Revision 1.17 2000/04/30 03:55:18 robertj
* Improved the RTCP messages, epecially reports
*
* Revision 1.16 2000/04/10 17:39:21 robertj
* Fixed debug output of RTP payload types to allow for unknown numbers.
*
* Revision 1.15 2000/04/05 03:17:31 robertj
* Added more RTP statistics gathering and H.245 round trip delay calculation.
*
* Revision 1.14 2000/03/23 02:55:18 robertj
* Added sending of SDES control packets.
*
* Revision 1.13 2000/03/20 20:54:04 robertj
* Fixed problem with being able to reopen for reading an RTP_Session (Cisco compatibilty)
*
* Revision 1.12 2000/02/29 13:00:13 robertj
* Added extra statistic display for RTP packets out of order.
*
* Revision 1.11 1999/12/30 09:14:31 robertj
* Changed payload type functions to use enum.
*
* Revision 1.10 1999/12/23 23:02:35 robertj
* File reorganision for separating RTP from H.323 and creation of LID for VPB support.
*
* Revision 1.9 1999/11/20 05:35:57 robertj
* Fixed possibly I/O block in RTP read loops.
*
* Revision 1.8 1999/11/19 09:17:15 robertj
* Fixed problems with aycnhronous shut down of logical channels.
*
* Revision 1.7 1999/11/14 11:41:18 robertj
* Added access functions to RTP statistics.
*
* Revision 1.6 1999/09/08 04:05:48 robertj
* Added support for video capabilities & codec, still needs the actual codec itself!
*
* Revision 1.5 1999/08/31 12:34:18 robertj
* Added gatekeeper support.
*
* Revision 1.4 1999/07/13 09:53:24 robertj
* Fixed some problems with jitter buffer and added more debugging.
*
* Revision 1.3 1999/07/09 06:09:49 robertj
* Major implementation. An ENORMOUS amount of stuff added everywhere.
*
* Revision 1.2 1999/06/22 13:49:40 robertj
* Added GSM support and further RTP protocol enhancements.
*
* Revision 1.1 1999/06/14 06:12:25 robertj
* Changes for using RTP sessions correctly in H323 Logical Channel context
*
*/
#ifndef __OPAL_RTP_H
#define __OPAL_RTP_H
#ifdef P_USE_PRAGMA
#pragma interface
#endif
#include <openh323buildopts.h>
#include <ptlib/sockets.h>
class RTP_JitterBuffer;
class PHandleAggregator;
#ifdef P_STUN
class PNatMethod;
#endif
///////////////////////////////////////////////////////////////////////////////
//
// class to hold the QoS definitions for an RTP channel
#if P_HAS_QOS
class RTP_QOS : public PObject
{
PCLASSINFO(RTP_QOS,PObject);
public:
PQoS dataQoS;
PQoS ctrlQoS;
};
#else
class RTP_QOS;
#endif
///////////////////////////////////////////////////////////////////////////////
// Real Time Protocol - IETF RFC1889 and RFC1890
/**An RTP data frame encapsulation.
*/
class RTP_DataFrame : public PBYTEArray
{
PCLASSINFO(RTP_DataFrame, PBYTEArray);
public:
RTP_DataFrame(PINDEX payloadSize = 2048, PBoolean dynamicAllocation = TRUE);
enum {
ProtocolVersion = 2,
MinHeaderSize = 12
};
enum PayloadTypes {
PCMU, // G.711 u-Law
FS1016, // Federal Standard 1016 CELP
G721, // ADPCM - Subsumed by G.726
G726 = G721,
GSM, // GSM 06.10
G7231, // G.723.1 at 6.3kbps or 5.3 kbps
DVI4_8k, // DVI4 at 8kHz sample rate
DVI4_16k, // DVI4 at 16kHz sample rate
LPC, // LPC-10 Linear Predictive CELP
PCMA, // G.711 A-Law
G722, // G.722
L16_Stereo, // 16 bit linear PCM
L16_Mono, // 16 bit linear PCM
G723, // G.723
CN, // Confort Noise
MPA, // MPEG1 or MPEG2 audio
G728, // G.728 16kbps CELP
DVI4_11k, // DVI4 at 11kHz sample rate
DVI4_22k, // DVI4 at 22kHz sample rate
G729, // G.729 8kbps
Cisco_CN, // Cisco systems comfort noise (unofficial)
CelB = 25, // Sun Systems Cell-B video
JPEG, // Motion JPEG
H261 = 31, // H.261
MPV, // MPEG1 or MPEG2 video
MP2T, // MPEG2 transport system
H263, // H.263
LastKnownPayloadType,
DynamicBase = 96,
MaxPayloadType = 127,
IllegalPayloadType
};
unsigned GetVersion() const { return (theArray[0]>>6)&3; }
PBoolean GetExtension() const { return (theArray[0]&0x10) != 0; }
void SetExtension(PBoolean ext);
PBoolean GetMarker() const { return (theArray[1]&0x80) != 0; }
void SetMarker(PBoolean m);
PayloadTypes GetPayloadType() const { return (PayloadTypes)(theArray[1]&0x7f); }
void SetPayloadType(PayloadTypes t);
WORD GetSequenceNumber() const { return *(PUInt16b *)&theArray[2]; }
void SetSequenceNumber(WORD n) { *(PUInt16b *)&theArray[2] = n; }
DWORD GetTimestamp() const { return *(PUInt32b *)&theArray[4]; }
void SetTimestamp(DWORD t) { *(PUInt32b *)&theArray[4] = t; }
DWORD GetSyncSource() const { return *(PUInt32b *)&theArray[8]; }
void SetSyncSource(DWORD s) { *(PUInt32b *)&theArray[8] = s; }
PINDEX GetContribSrcCount() const { return theArray[0]&0xf; }
DWORD GetContribSource(PINDEX idx) const;
void SetContribSource(PINDEX idx, DWORD src);
PINDEX GetHeaderSize() const;
int GetExtensionType() const; // -1 is no extension
void SetExtensionType(int type);
PINDEX GetExtensionSize() const;
PBoolean SetExtensionSize(PINDEX sz);
BYTE * GetExtensionPtr() const;
PINDEX GetPayloadSize() const { return payloadSize; }
PBoolean SetPayloadSize(PINDEX sz);
BYTE * GetPayloadPtr() const { return (BYTE *)(theArray+GetHeaderSize()); }
protected:
PINDEX payloadSize;
#if PTRACING
friend ostream & operator<<(ostream & o, PayloadTypes t);
#endif
};
PLIST(RTP_DataFrameList, RTP_DataFrame);
/**An RTP control frame encapsulation.
*/
class RTP_ControlFrame : public PBYTEArray
{
PCLASSINFO(RTP_ControlFrame, PBYTEArray);
public:
RTP_ControlFrame(PINDEX compoundSize = 2048);
unsigned GetVersion() const { return (BYTE)theArray[compoundOffset]>>6; }
unsigned GetCount() const { return (BYTE)theArray[compoundOffset]&0x1f; }
void SetCount(unsigned count);
enum PayloadTypes {
e_SenderReport = 200,
e_ReceiverReport,
e_SourceDescription,
e_Goodbye,
e_ApplDefined
};
unsigned GetPayloadType() const { return (BYTE)theArray[compoundOffset+1]; }
void SetPayloadType(unsigned t);
PINDEX GetPayloadSize() const { return 4*(*(PUInt16b *)&theArray[compoundOffset+2]); }
void SetPayloadSize(PINDEX sz);
BYTE * GetPayloadPtr() const { return (BYTE *)(theArray+compoundOffset+4); }
PBoolean ReadNextCompound();
PBoolean WriteNextCompound();
PINDEX GetCompoundSize() const { return compoundSize; }
#pragma pack(1)
struct ReceiverReport {
PUInt32b ssrc; /* data source being reported */
BYTE fraction; /* fraction lost since last SR/RR */
BYTE lost[3]; /* cumulative number of packets lost (signed!) */
PUInt32b last_seq; /* extended last sequence number received */
PUInt32b jitter; /* interarrival jitter */
PUInt32b lsr; /* last SR packet from this source */
PUInt32b dlsr; /* delay since last SR packet */
unsigned GetLostPackets() const { return (lost[0]<<16U)+(lost[1]<<8U)+lost[2]; }
void SetLostPackets(unsigned lost);
};
struct SenderReport {
PUInt32b ssrc; /* source this RTCP packet refers to */
PUInt32b ntp_sec; /* NTP timestamp */
PUInt32b ntp_frac;
PUInt32b rtp_ts; /* RTP timestamp */
PUInt32b psent; /* packets sent */
PUInt32b osent; /* octets sent */
};
enum DescriptionTypes {
e_END,
e_CNAME,
e_NAME,
e_EMAIL,
e_PHONE,
e_LOC,
e_TOOL,
e_NOTE,
e_PRIV,
NumDescriptionTypes
};
struct SourceDescription {
PUInt32b src; /* first SSRC/CSRC */
struct Item {
BYTE type; /* type of SDES item (enum DescriptionTypes) */
BYTE length; /* length of SDES item (in octets) */
char data[1]; /* text, not zero-terminated */
const Item * GetNextItem() const { return (const Item *)((char *)this + length + 2); }
Item * GetNextItem() { return (Item *)((char *)this + length + 2); }
} item[1]; /* list of SDES items */
};
SourceDescription & AddSourceDescription(
DWORD src ///< SSRC/CSRC identifier
);
SourceDescription::Item & AddSourceDescriptionItem(
SourceDescription & sdes, ///< SDES record to add item to
unsigned type, ///< Description type
const PString & data ///< Data for description
);
#pragma pack()
protected:
PINDEX compoundOffset;
PINDEX compoundSize;
};
class RTP_Session;
/**This class is the base for user data that may be attached to the RTP_session
allowing callbacks for statistics and progress monitoring to be passed to an
arbitrary object that an RTP consumer may require.
*/
class RTP_UserData : public PObject
{
PCLASSINFO(RTP_UserData, PObject);
public:
/**Callback from the RTP session for transmit statistics monitoring.
This is called every RTP_Session::txStatisticsInterval packets on the
transmitter indicating that the statistics have been updated.
The default behaviour does nothing.
*/
virtual void OnTxStatistics(
const RTP_Session & session ///< Session with statistics
) const;
/**Callback from the RTP session for receive statistics monitoring.
This is called every RTP_Session::receiverReportInterval packets on the
receiver indicating that the statistics have been updated.
The default behaviour does nothing.
*/
virtual void OnRxStatistics(
const RTP_Session & session ///< Session with statistics
) const;
/**Callback from the RTP session for statistics monitoring.
This is called at the end of a call to indicating
that the statistics of the call.
The default behaviour does nothing.
*/
virtual void OnFinalStatistics(
const RTP_Session & session ///< Session with statistics
) const;
};
/**This class is for encpsulating the IETF Real Time Protocol interface.
*/
class RTP_UDP;
class RTP_Session : public PObject
{
PCLASSINFO(RTP_Session, PObject);
public:
enum {
DefaultAudioSessionID = 1,
DefaultVideoSessionID = 2,
DefaultFaxSessionID = 3,
DefaultH224SessionID = 4,
DefaultExtVideoSessionID = 5
};
/**@name Construction */
//@{
/**Create a new RTP session.
*/
RTP_Session(
#ifdef H323_RTP_AGGREGATE
PHandleAggregator * aggregator, ///< RTP aggregator
#endif
unsigned id, ///< Session ID for RTP channel
RTP_UserData * userData = NULL ///< Optional data for session.
);
/**Delete a session.
This deletes the userData field.
*/
~RTP_Session();
//@}
/**@name Operations */
//@{
/**Sets the size of the jitter buffer to be used by this RTP session.
A session default to not having any jitter buffer enabled for reading
and the ReadBufferedData() function simply calls ReadData(). Once a
jitter buffer has been created it cannot be removed, though its size
may be adjusted.
If the jitterDelay paramter is zero, it destroys the jitter buffer
attached to this RTP session.
*/
void SetJitterBufferSize(
unsigned minJitterDelay, ///< Minimum jitter buffer delay in RTP timestamp units
unsigned maxJitterDelay, ///< Maximum jitter buffer delay in RTP timestamp units
PINDEX stackSize = 30000 ///< Stack size for jitter thread
);
/**Get current size of the jitter buffer.
This returns the currently used jitter buffer delay in RTP timestamp
units. It will be some value between the minimum and maximum set in
the SetJitterBufferSize() function.
*/
unsigned GetJitterBufferSize() const;
/**Modifies the QOS specifications for this RTP session*/
virtual PBoolean ModifyQOS(RTP_QOS * )
{ return FALSE; }
/**Read a data frame from the RTP channel.
This function will conditionally read data from eth jitter buffer or
directly if there is no jitter buffer enabled. An application should
generally use this in preference to directly calling ReadData().
*/
PBoolean ReadBufferedData(
DWORD timestamp, ///< Timestamp to read from buffer.
RTP_DataFrame & frame ///< Frame read from the RTP session
);
/**Read a data frame from the RTP channel.
Any control frames received are dispatched to callbacks and are not
returned by this function. It will block until a data frame is
available or an error occurs.
*/
virtual PBoolean ReadData(
RTP_DataFrame & frame, ///< Frame read from the RTP session
PBoolean loop ///< If TRUE, loop as long as data is available, if FALSE, only process once
) = 0;
/**Write a data frame from the RTP channel.
*/
virtual PBoolean WriteData(
RTP_DataFrame & frame ///< Frame to write to the RTP session
) = 0;
/**Write a control frame from the RTP channel.
*/
virtual PBoolean WriteControl(
RTP_ControlFrame & frame ///< Frame to write to the RTP session
) = 0;
/**Write the RTCP reports.
*/
virtual PBoolean SendReport();
/**Close down the RTP session.
*/
virtual void Close(
PBoolean reading ///< Closing the read side of the session
) = 0;
/**Get the local host name as used in SDES packes.
*/
virtual PString GetLocalHostName() = 0;
//@}
/**@name Call back functions */
//@{
enum SendReceiveStatus {
e_ProcessPacket,
e_IgnorePacket,
e_AbortTransport
};
virtual SendReceiveStatus OnSendData(RTP_DataFrame & frame);
virtual SendReceiveStatus OnReceiveData(const RTP_DataFrame & frame, const RTP_UDP & rtp);
virtual SendReceiveStatus OnReceiveControl(RTP_ControlFrame & frame);
class ReceiverReport : public PObject {
PCLASSINFO(ReceiverReport, PObject);
public:
void PrintOn(ostream &) const;
DWORD sourceIdentifier;
DWORD fractionLost; /* fraction lost since last SR/RR */
DWORD totalLost; /* cumulative number of packets lost (signed!) */
DWORD lastSequenceNumber; /* extended last sequence number received */
DWORD jitter; /* interarrival jitter */
PTimeInterval lastTimestamp;/* last SR packet from this source */
PTimeInterval delay; /* delay since last SR packet */
};
PARRAY(ReceiverReportArray, ReceiverReport);
class SenderReport : public PObject {
PCLASSINFO(SenderReport, PObject);
public:
void PrintOn(ostream &) const;
DWORD sourceIdentifier;
PTime realTimestamp;
DWORD rtpTimestamp;
DWORD packetsSent;
DWORD octetsSent;
};
virtual void OnRxSenderReport(const SenderReport & sender,
const ReceiverReportArray & reports);
virtual void OnRxReceiverReport(DWORD src,
const ReceiverReportArray & reports);
class SourceDescription : public PObject {
PCLASSINFO(SourceDescription, PObject);
public:
SourceDescription(DWORD src) { sourceIdentifier = src; }
void PrintOn(ostream &) const;
DWORD sourceIdentifier;
POrdinalToString items;
};
PARRAY(SourceDescriptionArray, SourceDescription);
virtual void OnRxSourceDescription(const SourceDescriptionArray & descriptions);
virtual void OnRxGoodbye(const PDWORDArray & sources,
const PString & reason);
virtual void OnRxApplDefined(const PString & type, unsigned subtype, DWORD src,
const BYTE * data, PINDEX size);
//@}
/**@name Member variable access */
//@{
/**Get the ID for the RTP session.
*/
unsigned GetSessionID() const { return sessionID; }
/**Get the canonical name for the RTP session.
*/
PString GetCanonicalName() const;
/**Set the canonical name for the RTP session.
*/
void SetCanonicalName(const PString & name);
/**Get the tool name for the RTP session.
*/
PString GetToolName() const;
/**Set the tool name for the RTP session.
*/
void SetToolName(const PString & name);
/**Get the user data for the session.
*/
RTP_UserData * GetUserData() const { return userData; }
/**Set the user data for the session.
*/
void SetUserData(
RTP_UserData * data ///< New user data to be used
);
/**Get the source output identifier.
*/
DWORD GetSyncSourceOut() const { return syncSourceOut; }
/**Increment reference count for RTP session.
*/
void IncrementReference() { referenceCount++; }
/**Decrement reference count for RTP session.
*/
PBoolean DecrementReference() { return --referenceCount == 0; }
/**Indicate if will ignore all but first received SSRC value.
*/
PBoolean WillIgnoreOtherSources() const { return ignoreOtherSources; }
/**Indicate if will ignore all but first received SSRC value.
*/
void SetIgnoreOtherSources(
PBoolean ignore ///< Flag for ignore other SSRC values
) { ignoreOtherSources = ignore; }
/**Indicate if will ignore out of order packets.
*/
PBoolean WillIgnoreOutOfOrderPackets() const { return ignoreOutOfOrderPackets; }
/**Indicate if will ignore out of order packets.
*/
void SetIgnoreOutOfOrderPackets(
PBoolean ignore ///< Flag for ignore out of order packets
) { ignoreOutOfOrderPackets = ignore; }
/**Get the time interval for sending RTCP reports in the session.
*/
const PTimeInterval & GetReportTimeInterval() { return reportTimeInterval; }
/**Set the time interval for sending RTCP reports in the session.
*/
void SetReportTimeInterval(
const PTimeInterval & interval ///< New time interval for reports.
) { reportTimeInterval = interval; }
/**Get the current report timer
*/
PTimeInterval GetReportTimer()
{ return reportTimer; }
/**Get the interval for transmitter statistics in the session.
*/
unsigned GetTxStatisticsInterval() { return txStatisticsInterval; }
/**Set the interval for transmitter statistics in the session.
*/
void SetTxStatisticsInterval(
unsigned packets ///< Number of packets between callbacks
);
/**Get the interval for receiver statistics in the session.
*/
unsigned GetRxStatisticsInterval() { return rxStatisticsInterval; }
/**Set the interval for receiver statistics in the session.
*/
void SetRxStatisticsInterval(
unsigned packets ///< Number of packets between callbacks
);
/**Get total number of packets sent in session.
*/
DWORD GetPacketsSent() const { return packetsSent; }
/**Get total number of octets sent in session.
*/
DWORD GetOctetsSent() const { return octetsSent; }
/**Get total number of packets received in session.
*/
DWORD GetPacketsReceived() const { return packetsReceived; }
/**Get total number of octets received in session.
*/
DWORD GetOctetsReceived() const { return octetsReceived; }
/**Get total number received packets lost in session.
*/
DWORD GetPacketsLost() const { return packetsLost; }
/**Get total number of packets received out of order in session.
*/
DWORD GetPacketsOutOfOrder() const { return packetsOutOfOrder; }
/**Get total number received packets too late to go into jitter buffer.
*/
DWORD GetPacketsTooLate() const;
/**Get average time between sent packets.
This is averaged over the last txStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetAverageSendTime() const { return averageSendTime; }
/**Get maximum time between sent packets.
This is over the last txStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetMaximumSendTime() const { return maximumSendTime; }
/**Get minimum time between sent packets.
This is over the last txStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetMinimumSendTime() const { return minimumSendTime; }
/**Get average time between received packets.
This is averaged over the last rxStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetAverageReceiveTime() const { return averageReceiveTime; }
/**Get maximum time between received packets.
This is over the last rxStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetMaximumReceiveTime() const { return maximumReceiveTime; }
/**Get minimum time between received packets.
This is over the last rxStatisticsInterval packets and is in
milliseconds.
*/
DWORD GetMinimumReceiveTime() const { return minimumReceiveTime; }
/**Get averaged jitter time for received packets.
This is the calculated statistical variance of the interarrival
time of received packets in milliseconds.
*/
DWORD GetAvgJitterTime() const { return jitterLevel>>7; }
/**Get averaged jitter time for received packets.
This is the maximum value of jitterLevel for the session.
*/
DWORD GetMaxJitterTime() const { return maximumJitterLevel>>7; }
/**
* return the timestamp at which the first packet of RTP data was received
*/
PTime GetFirstDataReceivedTime() const { return firstDataReceivedTime; }
/**
* return the local Transport Address
*/
PString GetLocalTransportAddress() const { return localAddress; };
/**
* return the remote Transport Address
*/
PString GetRemoteTransportAddress() const { return remoteAddress; };
//@}
/**@name Functions added to RTP aggregator */
//@{
virtual int GetDataSocketHandle() const
{ return -1; }
virtual int GetControlSocketHandle() const
{ return -1; }
//@}
protected:
void AddReceiverReport(RTP_ControlFrame::ReceiverReport & receiver);
unsigned sessionID;
PString canonicalName;
PString toolName;
unsigned referenceCount;
RTP_UserData * userData;
#ifndef NO_H323_AUDIO_CODECS
RTP_JitterBuffer * jitter;
#endif
PBoolean ignoreOtherSources;
PBoolean ignoreOutOfOrderPackets;
DWORD syncSourceOut;
DWORD syncSourceIn;
PTimeInterval reportTimeInterval;
unsigned txStatisticsInterval;
unsigned rxStatisticsInterval;
WORD lastSentSequenceNumber;
WORD expectedSequenceNumber;
DWORD lastSentTimestamp;
PTimeInterval lastSentPacketTime;
PTimeInterval lastReceivedPacketTime;
WORD lastRRSequenceNumber;
PINDEX consecutiveOutOfOrderPackets;
// Statistics
DWORD packetsSent;
DWORD octetsSent;
DWORD packetsReceived;
DWORD octetsReceived;
DWORD packetsLost;
DWORD packetsOutOfOrder;
DWORD averageSendTime;
DWORD maximumSendTime;
DWORD minimumSendTime;
DWORD averageReceiveTime;
DWORD maximumReceiveTime;
DWORD minimumReceiveTime;
DWORD jitterLevel;
DWORD maximumJitterLevel;
// Socket information
PString localAddress;
PString remoteAddress;
unsigned txStatisticsCount;
unsigned rxStatisticsCount;
DWORD averageSendTimeAccum;
DWORD maximumSendTimeAccum;
DWORD minimumSendTimeAccum;
DWORD averageReceiveTimeAccum;
DWORD maximumReceiveTimeAccum;
DWORD minimumReceiveTimeAccum;
DWORD packetsLostSinceLastRR;
DWORD lastTransitTime;
PTime firstDataReceivedTime;
PMutex reportMutex;
PTimer reportTimer;
#ifdef H323_RTP_AGGREGATE
PHandleAggregator * aggregator;
#endif
};
/**This class is for encpsulating the IETF Real Time Protocol interface.
*/
class RTP_SessionManager : public PObject
{
PCLASSINFO(RTP_SessionManager, PObject);
public:
/**@name Construction */
//@{
/**Construct new session manager database.
*/
RTP_SessionManager();
RTP_SessionManager(const RTP_SessionManager & sm);
RTP_SessionManager & operator=(const RTP_SessionManager & sm);
//@}
/**@name Operations */
//@{
/**Use an RTP session for the specified ID.
If this function returns a non-null value, then the ReleaseSession()
function MUST be called or the session is never deleted for the
lifetime of the session manager.
If there is no session of the specified ID, then you MUST call the
AddSession() function with a new RTP_Session. The mutex flag is left
locked in this case. The AddSession() expects the mutex to be locked
and unlocks it automatically.
*/
RTP_Session * UseSession(
unsigned sessionID ///< Session ID to use.
);
/**Add an RTP session for the specified ID.
This function MUST be called only after the UseSession() function has
returned NULL. The mutex flag is left locked in that case. This
function expects the mutex to be locked and unlocks it automatically.
*/
void AddSession(
RTP_Session * session ///< Session to add.
);
/**Release the session. If the session ID is not being used any more any
clients via the UseSession() function, then the session is deleted.
*/
void ReleaseSession(
unsigned sessionID ///< Session ID to release.
);
/**Get a session for the specified ID.
Unlike UseSession, this does not increment the usage count on the
session so may be used to just gain a pointer to an RTP session.
*/
RTP_Session * GetSession(
unsigned sessionID /// Session ID to get.
) const;
/**Begin an enumeration of the RTP sessions.
This function acquires the mutex for the sessions database and returns
the first element in it.
eg:
RTP_Session * session;
for (session = rtpSessions.First(); session != NULL; session = rtpSessions.Next()) {
if (session->Something()) {
rtpSessions.Exit();
break;
}
}
Note that the Exit() function must be called if the enumeration is
stopped prior to Next() returning NULL.
*/
RTP_Session * First();
/**Get the next session in the enumeration.
The session database remains locked until this function returns NULL.
Note that the Exit() function must be called if the enumeration is
stopped prior to Next() returning NULL.
*/
RTP_Session * Next();
/**Exit the enumeration of RTP sessions.
If the enumeration is desired to be exited before Next() returns NULL
this this must be called to unlock the session database.
Note that you should NOT call Exit() if Next() HAS returned NULL, or
race conditions can result.
*/
void Exit();
//@}
protected:
PDICTIONARY(SessionDict, POrdinalKey, RTP_Session);
SessionDict sessions;
PMutex mutex;
PINDEX enumerationIndex;
};
/**This class is for the IETF Real Time Protocol interface on UDP/IP.
*/
class H323Connection;
class RTP_UDP : public RTP_Session
{
PCLASSINFO(RTP_UDP, RTP_Session);
public:
/**@name Construction */
//@{
/**Create a new RTP channel.
*/
RTP_UDP(
#ifdef H323_RTP_AGGREGATE
PHandleAggregator * aggregator, ///< RTP aggregator
#endif
unsigned id, ///< Session ID for RTP channel
PBoolean remoteIsNat = FALSE ///< If TRUE, we have hints the remote endpoint is behind a NAT
);
/// Destroy the RTP
~RTP_UDP();
//@}
/**@name Overrides from class RTP_Session */
//@{
/**Read a data frame from the RTP channel.
Any control frames received are dispatched to callbacks and are not
returned by this function. It will block until a data frame is
available or an error occurs.
*/
virtual PBoolean ReadData(RTP_DataFrame & frame, PBoolean loop);
/**Write a data frame from the RTP channel.
*/
virtual PBoolean WriteData(RTP_DataFrame & frame);
/**Write a control frame from the RTP channel.
*/
virtual PBoolean WriteControl(RTP_ControlFrame & frame);
/**Close down the RTP session.
*/
virtual void Close(
PBoolean reading ///< Closing the read side of the session
);
/**Get the session description name.
*/
virtual PString GetLocalHostName();
//@}
/**@name QoS Settings */
//@{
/**Change the QoS settings
*/
virtual PBoolean ModifyQOS(RTP_QOS * rtpqos);
/** Enable QOS on Call Basis
*/
virtual void EnableGQoS(PBoolean success = TRUE);
#if P_HAS_QOS
/** Get the QOS settings
*/
PQoS & GetQOS();
#endif
//@}
/**@name New functions for class */
//@{
/**Open the UDP ports for the RTP session.
*/
PBoolean Open(
PIPSocket::Address localAddress, ///< Local interface to bind to
WORD portBase, ///< Base of ports to search
WORD portMax, ///< end of ports to search (inclusive)
BYTE ipTypeOfService, ///< Type of Service byte
const H323Connection & connection, ///< Connection
#ifdef P_STUN
PNatMethod * meth = NULL, ///* Nat Method to use to create sockets (or NULL if no Method)
#else
void * = NULL, ///< STUN server to use createing sockets (or NULL if no STUN)
#endif
RTP_QOS * rtpqos = NULL ///< QOS spec (or NULL if no QoS)
);
//@}
/**Reopens an existing session in the given direction.
*/
void Reopen(PBoolean isReading);
//@}
/**@name Member variable access */
//@{
/**Get local address of session.
*/
PIPSocket::Address GetLocalAddress() const { return localAddress; }
/**Set local address of session.
*/
void SetLocalAddress(
const PIPSocket::Address & addr
) { localAddress = addr; }
/**Get remote address of session.
*/
PIPSocket::Address GetRemoteAddress() const { return remoteAddress; }
/**Get local data port of session.
*/
WORD GetLocalDataPort() const { return localDataPort; }
/**Get local control port of session.
*/
WORD GetLocalControlPort() const { return localControlPort; }
/**Get remote data port of session.
*/
WORD GetRemoteDataPort() const { return remoteDataPort; }
/**Get remote control port of session.
*/
WORD GetRemoteControlPort() const { return remoteControlPort; }
/**Get data UDP socket of session.
*/
PUDPSocket & GetDataSocket() { return *dataSocket; }
/**Get control UDP socket of session.
*/
PUDPSocket & GetControlSocket() { return *controlSocket; }
/**Set the remote address and port information for session.
*/
PBoolean SetRemoteSocketInfo(
PIPSocket::Address address, ///< Address of remote
WORD port, ///< Port on remote
PBoolean isDataPort ///< Flag for data or control channel
);
/**Apply QOS - requires address to connect the socket on Windows platforms
*/
void ApplyQOS(
const PIPSocket::Address & addr
);
//@}
int GetDataSocketHandle() const
{ return dataSocket != NULL ? dataSocket->GetHandle() : -1; }
int GetControlSocketHandle() const
{ return controlSocket != NULL ? controlSocket->GetHandle() : -1; }
protected:
SendReceiveStatus ReadDataPDU(RTP_DataFrame & frame);
SendReceiveStatus ReadControlPDU();
SendReceiveStatus ReadDataOrControlPDU(
PUDPSocket & socket,
PBYTEArray & frame,
PBoolean fromDataChannel
);
PIPSocket::Address localAddress;
WORD localDataPort;
WORD localControlPort;
PIPSocket::Address remoteAddress;
WORD remoteDataPort;
WORD remoteControlPort;
PIPSocket::Address remoteTransmitAddress;
PBoolean shutdownRead;
PBoolean shutdownWrite;
PUDPSocket * dataSocket;
PUDPSocket * controlSocket;
PBoolean appliedQOS;
PBoolean enableGQOS;
PBoolean remoteIsNAT;
};
#endif // __OPAL_RTP_H
/////////////////////////////////////////////////////////////////////////////
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