diff --git a/sdrbase/audio/audionetsink.cpp b/sdrbase/audio/audionetsink.cpp
index 6bc9ff201..275539c81 100644
--- a/sdrbase/audio/audionetsink.cpp
+++ b/sdrbase/audio/audionetsink.cpp
@@ -269,6 +269,19 @@ 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:
         {
@@ -314,6 +327,25 @@ 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);
@@ -359,6 +391,21 @@ 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];
@@ -390,6 +437,27 @@ 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 ed4ce92f7..3ebe842a5 100644
--- a/sdrbase/util/rtpsink.cpp
+++ b/sdrbase/util/rtpsink.cpp
@@ -112,9 +112,9 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
         break;
     case PayloadOpus:
         m_sampleBytes = 1;
-        m_rtpSession.SetDefaultPayloadType(101);
-        m_packetSamples = 960; // Fixed 20ms @ 48 kHz as per https://tools.ietf.org/html/rfc7587
-        timestampinc = 960;    // and per single channel
+        m_rtpSession.SetDefaultPayloadType(96);
+        m_packetSamples = 160; // Payload size is 160 bytes
+        timestampinc = 960;    // But increment is 960
         break;
     case PayloadL16Mono:
     default:
@@ -336,6 +336,9 @@ unsigned int RTPSink::elemLength(PayloadType payloadType)
     {
     case PayloadPCMA8:
     case PayloadPCMU8:
+    case PayloadG722:
+    case PayloadOpus:
+    case PayloadL8:
         return sizeof(int8_t);
         break;
     case PayloadL16Stereo: