diff --git a/sdrbase/audio/audionetsink.cpp b/sdrbase/audio/audionetsink.cpp
index d0dbb372c..7de15b3ee 100644
--- a/sdrbase/audio/audionetsink.cpp
+++ b/sdrbase/audio/audionetsink.cpp
@@ -15,6 +15,8 @@
 // along with this program. If not, see <http://www.gnu.org/licenses/>.          //
 ///////////////////////////////////////////////////////////////////////////////////
 
+#include <algorithm>
+
 #include "audionetsink.h"
 #include "util/rtpsink.h"
 
@@ -33,7 +35,7 @@ AudioNetSink::AudioNetSink(QObject *parent) :
     m_bufferIndex(0),
     m_port(9998)
 {
-    memset(m_data, 0, 65536);
+    std::fill(m_data, m_data+m_dataBlockSize, 0);
     m_udpSocket = new QUdpSocket(parent);
 }
 
@@ -47,7 +49,7 @@ AudioNetSink::AudioNetSink(QObject *parent, int sampleRate, bool stereo) :
     m_bufferIndex(0),
     m_port(9998)
 {
-    memset(m_data, 0, 65536);
+    std::fill(m_data, m_data+m_dataBlockSize, 0);
     m_udpSocket = new QUdpSocket(parent);
     m_rtpBufferAudio = new RTPSink(m_udpSocket, sampleRate, stereo);
 }
@@ -130,6 +132,9 @@ void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
         case CodecL8:
             m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadL8, sampleRate);
             break;
+        case CodecG722:
+            m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadG722, sampleRate/2);
+            break;
         case CodecL16: // actually no codec
         default:
             m_rtpBufferAudio->setPayloadInformation(stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono, sampleRate);
@@ -173,35 +178,44 @@ void AudioNetSink::write(qint16 isample)
             m_udpSocket->writeDatagram((const char*)m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
             m_bufferIndex = 0;
         }
-        else
+
+        switch(m_codec)
         {
-            switch(m_codec)
-            {
-            case CodecPCMA:
-            case CodecPCMU:
-            {
-                qint8 *p = (qint8*) &m_data[m_bufferIndex];
-                *p = m_audioCompressor.compress8(sample);
-                m_bufferIndex += sizeof(qint8);
-            }
-                break;
-            case CodecL8:
-            {
-                qint8 *p = (qint8*) &m_data[m_bufferIndex];
-                *p = sample / 256;
-                m_bufferIndex += sizeof(qint8);
-            }
-                break;
-            case CodecL16:
-            default:
-            {
-                qint16 *p = (qint16*) &m_data[m_bufferIndex];
-                *p = sample;
-                m_bufferIndex += sizeof(qint16);
-            }
-                break;
+        case CodecPCMA:
+        case CodecPCMU:
+        {
+            qint8 *p = (qint8*) &m_data[m_bufferIndex];
+            *p = m_audioCompressor.compress8(sample);
+            m_bufferIndex += sizeof(qint8);
+        }
+            break;
+        case CodecL8:
+        {
+            qint8 *p = (qint8*) &m_data[m_bufferIndex];
+            *p = sample / 256;
+            m_bufferIndex += sizeof(qint8);
+        }
+            break;
+        case CodecG722:
+        {
+            qint16 *p = (qint16*) &m_data[m_udpBlockSize + 2*m_bufferIndex];
+            *p = sample;
+            m_bufferIndex += 1;
+
+            if (m_bufferIndex == m_udpBlockSize) {
+                m_g722.encode((uint8_t *) m_data, (const int16_t*) &m_data[3*m_udpBlockSize], m_udpBlockSize);
             }
         }
+            break;
+        case CodecL16:
+        default:
+        {
+            qint16 *p = (qint16*) &m_data[m_bufferIndex];
+            *p = sample;
+            m_bufferIndex += sizeof(qint16);
+        }
+            break;
+        }
     }
     else if (m_type == SinkRTP)
     {
@@ -220,6 +234,22 @@ void AudioNetSink::write(qint16 isample)
             m_rtpBufferAudio->write((uint8_t *) &p);
         }
             break;
+        case CodecG722:
+        {
+            if (m_bufferIndex >= m_g722BlockSize)
+            {
+                static const int sz = m_g722BlockSize / sizeof(int16_t);
+                uint8_t g722_data[sz];
+                m_g722.encode(g722_data, (const int16_t*) m_data, sz);
+                m_rtpBufferAudio->write(g722_data, sz);
+                m_bufferIndex = 0;
+            }
+
+            qint16 *p = (qint16*) &m_data[m_bufferIndex];
+            *p = sample;
+            m_bufferIndex += sizeof(qint16);
+        }
+            break;
         case CodecL16:
         default:
             m_rtpBufferAudio->write((uint8_t *) &sample);
diff --git a/sdrbase/audio/audionetsink.h b/sdrbase/audio/audionetsink.h
index 02db8e709..b60ebdaf4 100644
--- a/sdrbase/audio/audionetsink.h
+++ b/sdrbase/audio/audionetsink.h
@@ -21,6 +21,7 @@
 #include "dsp/dsptypes.h"
 #include "audiofilter.h"
 #include "audiocompressor.h"
+#include "audiog722.h"
 #include "export.h"
 
 #include <QObject>
@@ -44,7 +45,8 @@ public:
         CodecL16,  //!< Linear 16 bit samples (no formatting)
         CodecL8,   //!< Linear 8 bit samples
         CodecPCMA, //!< PCM A-law 8 bit samples
-        CodecPCMU  //!< PCM Mu-law 8 bit samples
+        CodecPCMU, //!< PCM Mu-law 8 bit samples
+        CodecG722  //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
     } Codec;
 
     AudioNetSink(QObject *parent); //!< without RTP
@@ -66,6 +68,8 @@ public:
     void moveToThread(QThread *thread);
 
     static const int m_udpBlockSize;
+    static const int m_dataBlockSize = 65536;
+    static const int m_g722BlockSize = 1024;  // size in bytes
 
 protected:
     SinkType m_type;
@@ -73,11 +77,12 @@ protected:
     QUdpSocket *m_udpSocket;
     RTPSink *m_rtpBufferAudio;
     AudioCompressor m_audioCompressor;
+    AudioG722 m_g722;
     AudioFilter m_audioFilter;
     int m_sampleRate;
     uint32_t m_decimation;
     uint32_t m_decimationCount;
-    char m_data[65536];
+    char m_data[m_dataBlockSize];
     unsigned int m_bufferIndex;
     QHostAddress m_address;
     unsigned int m_port;
diff --git a/sdrbase/audio/audiooutput.h b/sdrbase/audio/audiooutput.h
index 5c389c731..e5fea5926 100644
--- a/sdrbase/audio/audiooutput.h
+++ b/sdrbase/audio/audiooutput.h
@@ -46,7 +46,8 @@ public:
         UDPCodecL16,   //!< Linear 16 bit (no codec)
 		UDPCodecL8,    //!< Linear 8 bit
         UDPCodecALaw,  //!< PCM A-law 8 bit
-        UDPCodecULaw   //!< PCM Mu-law 8 bit
+        UDPCodecULaw,  //!< PCM Mu-law 8 bit
+        UDPCodecG722   //!< G722 compression
     };
 
 	AudioOutput();
diff --git a/sdrbase/util/rtpsink.cpp b/sdrbase/util/rtpsink.cpp
index 999b16679..9e92793c7 100644
--- a/sdrbase/util/rtpsink.cpp
+++ b/sdrbase/util/rtpsink.cpp
@@ -104,6 +104,12 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
         m_packetSamples = m_sampleRate / 50; // 20ms packet samples
         timestampinc = m_sampleRate / 100;   // 2 channels
         break;
+    case PayloadG722:
+        m_sampleBytes = 1;
+        m_rtpSession.SetDefaultPayloadType(9);
+        m_packetSamples = m_sampleRate / 50; // 20ms packet samples
+        timestampinc = m_sampleRate / 50;    // 1 channel
+        break;
     case PayloadL16Mono:
     default:
         m_sampleBytes = 2;
@@ -226,10 +232,10 @@ void RTPSink::write(const uint8_t *sampleByte)
             qCritical("RTPSink::write: cannot write packet: %s", qrtplib::RTPGetErrorString(status).c_str());
         }
 
-        writeNetBuf(&m_byteBuffer[0], 
-            sampleByte,  
-            elemLength(m_payloadType), 
-            m_sampleBytes, 
+        writeNetBuf(&m_byteBuffer[0],
+            sampleByte,
+            elemLength(m_payloadType),
+            m_sampleBytes,
             m_endianReverse);
         m_sampleBufferIndex = 1;
     }
diff --git a/sdrbase/util/rtpsink.h b/sdrbase/util/rtpsink.h
index ea69b4549..8f6a421ac 100644
--- a/sdrbase/util/rtpsink.h
+++ b/sdrbase/util/rtpsink.h
@@ -44,7 +44,8 @@ public:
         PayloadL16Stereo,
         PayloadL8,
         PayloadPCMA8,
-        PayloadPCMU8
+        PayloadPCMU8,
+        PayloadG722
     } PayloadType;
 
     RTPSink(QUdpSocket *udpSocket, int sampleRate, bool stereo);
diff --git a/sdrgui/gui/audiodialog.cpp b/sdrgui/gui/audiodialog.cpp
index 2b59d3af0..167184f84 100644
--- a/sdrgui/gui/audiodialog.cpp
+++ b/sdrgui/gui/audiodialog.cpp
@@ -1,10 +1,29 @@
-#include "audiodialog.h"
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2015-2019 F4EXB                                                 //
+// written by Edouard Griffiths                                                  //
+//                                                                               //
+// This program is free software; you can redistribute it and/or modify          //
+// it under the terms of the GNU General Public License as published by          //
+// the Free Software Foundation as version 3 of the License, or                  //
+//                                                                               //
+// This program is distributed in the hope that it will be useful,               //
+// but WITHOUT ANY WARRANTY; without even the implied warranty of                //
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the                  //
+// GNU General Public License V3 for more details.                               //
+//                                                                               //
+// You should have received a copy of the GNU General Public License             //
+// along with this program. If not, see <http://www.gnu.org/licenses/>.          //
+///////////////////////////////////////////////////////////////////////////////////
 
-#include <audio/audiodevicemanager.h>
-#include <QTreeWidgetItem>
-#include "ui_audiodialog.h"
 #include <math.h>
 
+#include <QTreeWidgetItem>
+#include <QMessageBox>
+
+#include "audio/audiodevicemanager.h"
+#include "audiodialog.h"
+#include "ui_audiodialog.h"
+
 AudioDialogX::AudioDialogX(AudioDeviceManager* audioDeviceManager, QWidget* parent) :
 	QDialog(parent),
 	ui(new Ui::AudioDialog),
@@ -263,26 +282,43 @@ void AudioDialogX::updateOutputDeviceInfo()
 void AudioDialogX::updateOutputSDPString()
 {
     QString format;
-
-    switch(m_outputDeviceInfo.udpChannelCodec)
-    {
-    case AudioOutput::UDPCodecALaw:
-        format = "PCMA";
-        break;
-    case AudioOutput::UDPCodecULaw:
-        format = "PCMU";
-        break;
-    case AudioOutput::UDPCodecL8:
-        format = "L8";
-        break;
-    case AudioOutput::UDPCodecL16:
-    default:
-        format = "L16";
-        break;
-    }
-
     int nChannels = m_outputDeviceInfo.udpChannelMode == AudioOutput::UDPChannelStereo ? 2 : 1;
     uint32_t decimationFactor = m_outputDeviceInfo.udpDecimationFactor == 0 ? 1 : m_outputDeviceInfo.udpDecimationFactor;
 
+    if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecALaw)
+    {
+        format = "PCMA";
+        if ((nChannels != 1) || (m_outputDeviceInfo.sampleRate/decimationFactor != 8000)) {
+            QMessageBox::information(this, tr("Message"), tr("PCMA must be 8000 Hz single channel"));
+        }
+    }
+    else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecULaw)
+    {
+        format = "PCMU";
+        if ((nChannels != 1) || (m_outputDeviceInfo.sampleRate/decimationFactor != 8000)) {
+            QMessageBox::information(this, tr("Message"), tr("PCMU must be 8000 Hz single channel"));
+        }
+    }
+    else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecG722)
+    {
+        format = "G722";
+        if ((nChannels != 1) || (m_outputDeviceInfo.sampleRate/decimationFactor != 16000)) {
+            QMessageBox::information(this, tr("Message"), tr("G722 must be 16000 Hz single channel"));
+        }
+    }
+    else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecL8)
+    {
+        format = "L8";
+    }
+    else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecL16)
+    {
+        format = "L16";
+    }
+    else
+    {
+        QMessageBox::information(this, tr("Message"), tr("Unknown codec"));
+        format = "UNK";
+    }
+
     ui->outputSDPText->setText(tr("%1/%2/%3").arg(format).arg(m_outputDeviceInfo.sampleRate/decimationFactor).arg(nChannels));
 }
diff --git a/sdrgui/gui/audiodialog.ui b/sdrgui/gui/audiodialog.ui
index aed980dff..e422da931 100644
--- a/sdrgui/gui/audiodialog.ui
+++ b/sdrgui/gui/audiodialog.ui
@@ -255,6 +255,11 @@
              <string>PCMU</string>
             </property>
            </item>
+           <item>
+            <property name="text">
+             <string>G722</string>
+            </property>
+           </item>
           </widget>
          </item>
          <item>