diff --git a/sdrbase/audio/audionetsink.cpp b/sdrbase/audio/audionetsink.cpp
index 275539c81..6bc9ff201 100644
--- a/sdrbase/audio/audionetsink.cpp
+++ b/sdrbase/audio/audionetsink.cpp
@@ -269,19 +269,6 @@ void AudioNetSink::write(qint16 isample)
             }
         }
             break;
-        case CodecOpus:
-        {
-            if (m_codecInputIndex == m_codecInputSize)
-            {
-                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
-                nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
-                m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
-                m_codecInputIndex = 0;
-            }
-
-            m_opusIn[m_codecInputIndex++] = sample;
-        }
-            break;
         case CodecL16:
         default:
         {
@@ -327,25 +314,6 @@ void AudioNetSink::write(qint16 isample)
             m_bufferIndex += 1;
         }
             break;
-        case CodecOpus:
-        {
-            if (m_codecInputIndex == m_codecInputSize)
-            {
-                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
-                if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
-                    qWarning("AudioNetSink::write: CodecOpus mono: unexpected output frame size: %d bytes", nbBytes);
-                }
-                m_bufferIndex = 0;
-                m_codecInputIndex = 0;
-            }
-
-            if (m_codecInputIndex % m_codecRatio == 0) {
-                m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
-            }
-
-            m_opusIn[m_codecInputIndex++] = sample;
-        }
-            break;
         case CodecL16:
         default:
             m_rtpBufferAudio->write((uint8_t *) &sample);
@@ -391,21 +359,6 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
         case CodecPCMU:
         case CodecG722:
             break; // mono modes - do nothing
-        case CodecOpus:
-        {
-            if (m_codecInputIndex == m_codecInputSize)
-            {
-                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
-                nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
-                m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
-                m_codecInputIndex = 0;
-            }
-
-            m_opusIn[2*m_codecInputIndex]   = lSample;
-            m_opusIn[2*m_codecInputIndex+1] = rSample;
-            m_codecInputIndex++;
-        }
-            break;
         case CodecL8:
         {
             qint8 *p = (qint8*) &m_data[m_bufferIndex];
@@ -437,27 +390,6 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
         case CodecPCMU:
         case CodecG722:
             break; // mono modes - do nothing
-        case CodecOpus:
-        {
-            if (m_codecInputIndex == m_codecInputSize)
-            {
-                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
-                if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
-                    qWarning("AudioNetSink::write: CodecOpus stereo: unexpected output frame size: %d bytes", nbBytes);
-                }
-                m_bufferIndex = 0;
-                m_codecInputIndex = 0;
-            }
-
-            if (m_codecInputIndex % m_codecRatio == 0) {
-                m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
-            }
-
-            m_opusIn[2*m_codecInputIndex]   = lSample;
-            m_opusIn[2*m_codecInputIndex+1] = rSample;
-            m_codecInputIndex++;
-        }
-            break;
         case CodecL8:
         {
             qint8 pl = lSample / 256;
diff --git a/sdrbase/util/rtpsink.cpp b/sdrbase/util/rtpsink.cpp
index af1e3a53a..ed4ce92f7 100644
--- a/sdrbase/util/rtpsink.cpp
+++ b/sdrbase/util/rtpsink.cpp
@@ -107,8 +107,8 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
     case PayloadG722:
         m_sampleBytes = 1;
         m_rtpSession.SetDefaultPayloadType(9);
-        m_packetSamples = m_sampleRate / 50; // 20ms packet samples
-        timestampinc = m_sampleRate / 50;    // 1 channel
+        m_packetSamples = 160; // Fixed 8 kB/s 20ms packet samples
+        timestampinc = 160;    // 1 channel
         break;
     case PayloadOpus:
         m_sampleBytes = 1;
diff --git a/sdrgui/gui/audiodialog.cpp b/sdrgui/gui/audiodialog.cpp
index d85276a62..ca1f6514a 100644
--- a/sdrgui/gui/audiodialog.cpp
+++ b/sdrgui/gui/audiodialog.cpp
@@ -353,7 +353,7 @@ void AudioDialogX::check()
     else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecOpus)
     {
         int effectiveSampleRate = m_outputDeviceInfo.sampleRate/decimationFactor;
-        if ((effectiveSampleRate != 48000) || (effectiveSampleRate != 24000) || (effectiveSampleRate != 16000) || (effectiveSampleRate != 12000)) {
+        if ((effectiveSampleRate != 48000) && (effectiveSampleRate != 24000) && (effectiveSampleRate != 16000) && (effectiveSampleRate != 12000)) {
             QMessageBox::information(this, tr("Message"), tr("Opus takes only 48, 24, 16 or 12 kHz sample rates"));
         }
     }