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PyPI · #3142 most downloaded on PyPI
An implementation of WebRTC and ORTC
Last release 2 months ago
13 Jul 2026
Release timing varies
gaps range from 2 weeks to 9 months
Nearly every release is documented
notes for 54 of 56 stable releases
Nothing withdrawn
no release was ever pulled
9 years old
56 releases · first in 2018
Add support for RTCConfiguration.alwaysNegotiateDataChannels.
Add support for RTCConfiguration.alwaysNegotiateDataChannels.
Use the same same ICE credentials for all media sections.
Harden SCTP packet parsing against chunks with an invalid length.
Relax versioned dependency on PyAV to allow version 17.x. This fixes a per-frame memory leak.
Allow ~aiortc.contrib.media.MediaRecorder to record audio and / or video to a WebM container.
video to a WebM container.
Restore support for calling RTCPeerConnection.addIceCandidate before RTCPeerConnection.setRemoteDescription.
Send a keyframe if an RTCP Full Instantaneous Resolution (FIR) feedback is received.
Support parsing SDP rtcp attribute without a host.
Use the "modern" SDP form "UDP/DTLS/SCTP" in offers to negotiate data channels.
Relax versioned dependency on PyAV to allow version 15.x and 16.x.
Add support for Python 3.14, drop end-of-life Python 3.9.
One column per quarter.
Add support for G.722 audio codec.
Add support for G.722 audio codec.
Handle undecodable VP8 packages as was done prior to release 1.12.0.
Limit the number of threads used for VP8 encoding as was done prior to release 1.12.0.
Allow calling RTCPeerConnection.setLocalDescription with no argument to implicitly create an offer or answer as needed.
Allow calling RTCPeerConnection.addIceCandidate with None argument to signal remote end-of-candidates.
Support creating offers without media or data channels.
Reject STUN URLs containing a "transport" query parameter.
Ensure the webcam example shuts down cleanly on KeyboardInterrupt.
Use PyAV to perform Opus + VP8 encoding and decoding. This means aiortc is now pure Python.
is now pure Python.
Allow configuring the media-bundling policy using RTCConfiguration.bundlePolicy.
Fix reversed track negotiated media ID (mid) when using multiple transceivers of the same kind.
Fix the RTCIceServer.urls typing to accept List[str] in addition to str.
Fix octet count and packet count overflow in RTCRtpSender when sending RTCP sender reports.
Fix RTCPeerConnection.addIceCandidate when handling candidates with and m-line index and no media ID.
Require pyOpenSSL 25.x and fix deprecation warnings.
Fix decoding of RTX retransmission packets.
Drop support for obsolete h264_omx codec.
Require PyAV 14.x to support recent FFmpeg versions.
Require pyee 13.x for better typing support.
Require pyOpenSSL 25.x and fix deprecation warnings.
Build wheels for Linux aarch64 again.
Build wheels for Linux aarch64 again.
Be more cautious when releasing RTCRtpSender's encoder.
Set correct codec for ~aiortc.contrib.media.MediaRecorder OGG output.
Add support for Python 3.13, drop end-of-life Python 3.8.
Add support for Python 3.13, drop end-of-life Python 3.8.
Stop building wheels for Linux aarch64 for now due to CI instability.
Add py.typed to indicate the package has typings, fix some annotations.
Avoid early wraparound of RTP sequences numbers which can break SRTP.
Add support for sha-384 and sha-512 DTLS certificate fingerprints.
Allow using PyAV 13.x.
Stop using the deprecated audioop standard library module.
Handle offers with active or passive DTLS setups.
Stop using the deprecated audioop standard library module.
Allow using PyAV 12.x.
Only send / receive RTP according to RTCRtpTransceiver.currentDirection.
Only send / receive RTP according to RTCRtpTransceiver.currentDirection.
Close the RTCPeerConnection if all DTLS transports are closed.
Free the encoder as soon as the RTCRtpSender stops to save memory.
Modernise JavaScript in server and webcam examples.
Fix utcnow() deprecation warning on Python 3.12.
Add support for GCM based SRTP protection profiles.
Reduce supported DTLS cipher list to avoid Client Hello fragmentation.
Fix utcnow() deprecation warning on Python 3.12.
Build wheels using Py_LIMITED_ABI to make them compatible with future Python versions.
Build wheels using Py_LIMITED_ABI to make them compatible with future Python versions.
Build wheels using opus 1.4 and vpx 1.13.1.
Use unique IDs for audio and video header extensions.
Allow ~aiortc.contrib.media.MediaRecorder to record audio from pulse.
Make H.264 send a full picture when picture loss occurs.
Make H.264 send a full picture when picture loss occurs.
Fix TURN over TCP by updating aioice to 0.9.0.
Make use of the ifaddr package instead of the unmaintained netifaces package.
Allow ~aiortc.contrib.media.MediaPlayer to send media without transcoding.
Build wheels for Python 3.11.
Allow ~aiortc.contrib.media.MediaPlayer to send media without transcoding.
Allow ~aiortc.contrib.media.MediaPlayer to specify a timeout when opening media.
Make RTCSctpTransport transmit packets sooner to reduce datachannel latency.
Refactor RTCDtlsTransport to use PyOpenSSL.
Make RTCPeerConnection log sent and received SDP when using verbose logging.
Fix a deprecation warning when using av >= 9.1.0.
Limit size of NACK reports to avoid excessive packet size.
Improve H.264 codec matching.
Determine video size from first frame received by ~aiortc.contrib.media.MediaRecorder.
Fix a deprecation warning when using av >= 9.1.0.
Tolerate STUN URLs containing a protocol querystring argument.
Build wheels for aarch64 on Linux.
Build wheels for aarch64 on Linux.
Adapt ~aiortc.contrib.media.MediaPlayer for PyAV 9.x.
Ensure H.264 produces B-frames by resetting picture type.
Build wheels for Python 3.10 and for arm64 on Mac.
Build wheels for Python 3.10 and for arm64 on Mac.
Build wheels against libvpx 1.10.
Add support for looping in ~aiortc.contrib.media.MediaPlayer.
Add unbuffered option to ~aiortc.contrib.media.MediaRelay.
Calculate audio energy and send in RTP header extension.
Fix a race condition in RTP sender/receiver shutdown.
Improve performance of H.264 bitstream splitting code.
Update imports for pyee version 9.x.
Fully switch to google-crc32c instead of crc32.
Drop support for Python 3.6.
Remove apprtc code as the service is no longer publicly hosted.
Add a clear error message when no common codec is found.
Add a clear error message when no common codec is found.
Replace the crc32 dependency with google-crc32c which offers a more liberal license.
Fix jitter buffer to avoid severe picture corruption under packet loss and send Picture Loss Indication (PLI) when needed.
send Picture Loss Indication (PLI) when needed.
Make H.264 encoder honour the bitrate from the bandwidth estimator.
Add support for hardware-accelerated H.264 encoding on Raspberry Pi 4 using the h264_omx codec.
Add ~aiortc.contrib.media.MediaRelay class to allow sending media tracks to multiple consumers.
Add RTCPeerConnection.connectionState property.
Add RTCPeerConnection.connectionState property.
Correctly detect RTCIceTransport "failed" state.
Correctly route RTP packets when there are multiple tracks of the same kind.
Use full module name to name loggers.
Defer adding remote candidates until after transport bundling to avoid unnecessary mDNS lookups.
unnecessary mDNS lookups.
Add support for resolving mDNS candidates.
Add support for resolving mDNS candidates.
Improve support for TURN, especially long-lived connections.
Make RTCPeerConnection.addIceCandidate a coroutine.
Make RTCPeerConnection.addIceCandidate a coroutine.
Make RTCIceTransport.addRemoteCandidate a coroutine.
Handle SSRC attributes in SDP containing a colon (#372).
Limit number of H.264 NALU per packet (#394, #426).
server make it possible to specify bind address (#347).
Provide binary wheels for Linux, Mac and Windows on PyPI.
Provide binary wheels for Linux, Mac and Windows on PyPI.
Add RTCSctpTransport.maxChannels property.
Add RTCSctpTransport.maxChannels property.
Recycle stream IDs (#256).
Correctly close data channel when SCTP is not established (#300).
Add add RTCRtpReceiver.track property (#298).
Fix a crash in AimdRateControl (#295).
Drop support for OpenSSL < 1.0.2.
Drop support for OpenSSL < 1.0.2.
apprtc fix handling of empty "candidate" message.
Fix a MediaPlayer crash when stopping one track of a multi-track file (#237, #274).
Fix a MediaPlayer error when stopping a track while waiting for the next frame.
Make RTCRtpSender resilient to exceptions raised by media stream tracks (#283).
Do not repeatedly send key frames after receiving a PLI.
Do not repeatedly send key frames after receiving a PLI.
Do not try to determine track ID if there is no Msid.
Accept a star in rtcp-fb attributes.
Peer connection ...............
Assign DTLS role based on the SDP negotiation, not the resolved ICE role.
When the peer is ICE lite, adopt the ICE controlling role, and do not use agressive nomination.
Do not close transport on setRemoteDescription if media and data are bundled.
Set RemoteStreamTrack.id based on the Msid.
Support alsa hardware output in MediaRecorder.
Add support for the ice-lite attribute.
Add support for receiving session-level ice-ufrag, ice-pwd and setup attributes.
Switch from attrs to standard Python dataclasses.
Use PEP-526 style variable annotations instead of comments.
* Drop support for Python 3.5. * Drop dependency on PyOpenSSL. * Use PyAV >= 7.0.0. * Add partial type hints.
Drop support for Python 3.5. * Drop dependency on PyOpenSSL. * Use PyAV >= 7.0.0. * Add partial type hints.
Display exception if data handler fails.
Display exception if data handler fails.
server and webcam : add playsinline attribute for iOS compatibility.
webcam : make it possible to play media from a file.
Use aioice >= 0.6.15 to not fail on mDNS candidates.
Use pyee version 6.x.
Call SSL_CTX_set_ecdh_auto for OpenSSL 1.0.2.
Call SSL_CTX_set_ecdh_auto for OpenSSL 1.0.2.
Correctly route REMB packets to the RTCRtpSender.
~aiortc.contrib.media.MediaPlayer : release resources (e.g. webcam) when the player stops.
~aiortc.contrib.signaling.ApprtcSignaling : make AppRTC signaling available for more examples.
datachannel-cli : make uvloop optional.
videostream-cli : animate the flag with a wave effect.
webcam : explicitly set frame rate to 30 fps for webcams.
Support out-of-band negotiation and custom channel id.
Support out-of-band negotiation and custom channel id.
Fix documentation build by installing crc32c instead of crcmod.
~aiortc.contrib.media.MediaPlayer : skip frames with no presentation timestamp (pts).
Do not raise congestion window when it is not fully utilized.
Do not raise congestion window when it is not fully utilized.
Fix Highest TSN Newly Acknowledged logic for striking lost chunks.
Do not limit congestion window to 120kB, limit burst size instead.
Skip RTX packets with an empty payload.
apprtc : make the initiator send messages using an HTTP POST instead of WebSocket.
janus : new example to connect to the Janus WebRTC server.
server : add cartoon effect to video transforms.
Do not use DTLSv1_get_timeout after DTLS handshake completes.
Do not use DTLSv1_get_timeout after DTLS handshake completes.
Add setter for RTCDataChannel.bufferedAmountLowThreshold.
Use crc32c package instead of crcmod, it provides better performance.
Improve parsing and serialization code performance.
Disable logging code if it is not used to improve performance.
Do not bomb if SRTP is received before DTLS handshake completes.
Do not bomb if SRTP is received before DTLS handshake completes.
Implement unordered delivery, so that the ordered option is honoured.
Implement partial reliability, so that the maxRetransmits and maxPacketLifeTime options are honoured.
Put all tracks in the same stream for now, fixes breakage introduced in 0.9.14.
Use case-insensitive comparison for codec names.
Use a=msid attribute in SDP instead of SSRC-level attributes.
server : make it possible to select unreliable mode for data channels.
server : print the round-trip time for data channel messages.
Log OpenSSL errors if the DTLS handshake fails.
Log OpenSSL errors if the DTLS handshake fails.
Fix DTLS handshake in server mode with OpenSSL < 1.1.0.
Add RTCRtpReceiver.getCapabilities and RTCRtpSender.getCapabilities.
Add RTCRtpReceiver.getSynchronizationSources.
Add RTCRtpTransceiver.setCodecPreferences.
server : make it possible to force audio codec.
server : shutdown cleanly on Chrome which lacks RTCRtpTransceiver.stop.
Emit a warning if the crcmod C extension is not present.
Emit a warning if the crcmod C extension is not present.
Support subsequent offer / answer exchanges.
Route RTCP parameters to RTP receiver and sender independently.
Fix a regression when the remote SSRC are not known.
Fix VP8 descriptor parsing errors detected by fuzzing.
Fix H264 descriptor parsing errors detected by fuzzing.
Add support for RTX retransmission packets.
Add support for RTX retransmission packets.
Fix RTP and RTCP parsing errors detected by fuzzing.
Use case-insensitive comparison for hash algorithm in SDP, fixes interoperability with Asterisk.
Offer NACK PLI and REMB feedback mechanisms for H.264.
Raise an exception if RTCDataChannel.send is called when readyState is not 'open'.
Raise an exception if RTCDataChannel.send is called when readyState is not 'open'.
Do not use stream sequence number for unordered data channels.
Set VP8 target bitrate according to Receiver Estimated Maximum Bandwidth.
Correctly handle encoding in copy-and-paste signaling.
server : add command line options to use HTTPS.
webcam : add command line options to use HTTPS.
webcam : add code to open webcam on OS X.
Rework code in order to facilitate garbage collection and avoid memory leaks.
Rework code in order to facilitate garbage collection and avoid memory leaks.
Make AudioStreamTrack and VideoStreamTrack produce empty frames more regularly.
Make AudioStreamTrack and VideoStreamTrack produce empty frames more regularly.
Fix a regession in copy-and-paste signaling which blocked the event loop.
Peer connection ...............
Send raddr and rport parameters for server reflexive and relayed candidates. This is required for Firefox to accept our STUN / TURN candidates.
Do not raise an exception if ICE or DTLS connection fails, just change state.
Revert to using asyncio's run_in_executor to send data to the encoder, it greatly reduces the response time.
Adjust package requirements to accept PyAV < 7.0.0.
webcam : force Chrome to use "unified-plan" semantics to enabled addTransceiver.
~aiortc.contrib.media.MediaPlayer : don't sleep at all when playing from webcam. This eliminates the constant one-second lag in the webcam demo.
aiortc now uses PyAV's ~av.audio.frame.AudioFrame and ~av.video.frame.VideoFrame classes instead of defining its own.
Warning
aiortc now uses PyAV's ~av.audio.frame.AudioFrame and ~av.video.frame.VideoFrame classes instead of defining its own.
Use a jitter buffer for incoming audio.
Add RTCPeerConnection.addTransceiver method.
Add RTCRtpTransceiver.direction to manage transceiver direction.
apprtc : demonstrate the use of ~aiortc.contrib.media.MediaPlayer and ~aiortc.contrib.media.MediaRecorder.
webcam : new examples illustrating sending video from a webcam to a browser.
~aiortc.contrib.media.MediaPlayer : don't sleep if a frame lacks timing information.
~aiortc.contrib.media.MediaPlayer : remove start() and stop() methods.
~aiortc.contrib.media.MediaRecorder : use libx264 for encoding.
~aiortc.contrib.media.MediaRecorder : make start() and stop() coroutines.
Add support for H.264 video, a big thank you to @dsvictor94!
Add support for H.264 video, a big thank you to @dsvictor94!
Add support for sending Receiver Estimate Maximum Bitrate (REMB) feedback.
Add support for parsing / serializing more RTP header extensions.
Move each media encoder / decoder its one thread instead of using a thread pool.
Add the RTCPeerConnection.getStats() coroutine to retrieve statistics.
Add initial RTCTransportStats to report transport statistics.
Add new ~aiortc.contrib.media.MediaPlayer class to read audio / video from a file.
Add new ~aiortc.contrib.media.MediaRecorder class to write audio / video to a file.
Add new ~aiortc.contrib.media.MediaBlackhole class to discard audio / video.
Make RemoteStreamTrack emit an "ended" event, to simplify shutting down media consumers.
Make RemoteStreamTrack emit an "ended" event, to simplify shutting down media consumers.
Add RemoteStreamTrack.readyState property.
Handle timestamp wraparound on sent RTP packets.
Add a versioned dependency on cffi>=1.0.0 to fix Raspberry Pi builds.
Optimize reception for improved latency and throughput.
Optimize reception for improved latency and throughput.
Add initial RTCRtpReceiver.getStats() and RTCRtpReceiver.getStats() coroutines.
datachannel-cli: display ping/pong roundtrip time.
Make it possible to add multiple audio or video streams.
Make it possible to add multiple audio or video streams.
Implement basic RTP video packet loss detection / retransmission using RTCP NACK feedback.
Respond to Picture Loss Indications (PLI) by sending a keyframe.
Use shorter MID values to reduce RTP header extension overhead.
Correctly shutdown and discard unused transports when using BUNDLE.
server : make it possible to save received video to an AVI file.
Peer connection ...............
Add support for TURN over TCP.
Add media and signaling helpers in aiortc.contrib.
Fix colorspace OpenCV colorspace conversions.
apprtc : send rotating image on video track.
Set PictureID attribute on outgoing VP8 frames.
Set PictureID attribute on outgoing VP8 frames.
Negotiate and send SDES MID header extension for RTP packets.
Fix negative packets_lost encoding for RTCP reports.
Numerous performance improvements in congestion control.
Numerous performance improvements in congestion control.
datachannel-filexfer: use uvloop instead of default asyncio loop.
Revert making RTCDataChannel.send a coroutine.
Revert making RTCDataChannel.send a coroutine.
BREAKING CHANGE: make RTCDataChannel.send a coroutine.
Enable post-processing in VP8 decoder to remove (macro) blocks.
Set target bitrate for VP8 encoder to 900kbps.
Re-create VP8 encoder if frame size changes.
Implement jitter estimation for RTCP reports.
Avoid overflowing the DLSR field for RTCP reports.
Raise video jitter buffer size.
BREAKING CHANGE: make RTCDataChannel.send a coroutine.
Support spec-compliant SDP format for datachannels, as used in Firefox 63.
Never send a negative advertised_cwnd.
datachannel-filexfer: new example for file transfer over a data channel.
datachannel-vpn: new example for a VPN over a data channel.
server: make it possible to select video resolution.
Align VP8 settings with those used by WebRTC project, which greatly improves video quality.
Align VP8 settings with those used by WebRTC project, which greatly improves video quality.
Send RTCP source description, sender report, receiver report and bye packets.
server:
make it possible to not transform video at all.
allow video display to be up to 1280px wide.
videostream-cli:
fix Python 3.5 compatibility
Delay logging string interpolation to reduce cost of packet logging in non-verbose mode.
Peer connection ...............
Add RTCPeerConnection.addIceCandidate() method to handle trickled ICE candidates.
Make stop() methods of RTCRtpReceiver, RTCRtpSender and RTCRtpTransceiver coroutines to enable clean shutdown.
Clean up RTCDataChannel shutdown sequence.
Support receiving an SCTP RE-CONFIG to raise number of inbound streams.
server:
perform some image processing using OpenCV.
make it possible to disable data channels.
make demo web interface more mobile-friendly.
apprtc:
automatically create a room if no room is specified on command line.
handle bye command.
Peer connection ...............
Make it possible to specify one STUN server and / or one TURN server.
Add BUNDLE support to use a single ICE/DTLS transport for multiple media.
Move media encoding / decoding off the main thread.
Use SCTP ABORT instead of SHUTDOWN when stopping RTCSctpTransport.
Advertise support for SCTP RE-CONFIG extension.
Make RTCDataChannel emit open and close events.
Add an example of how to connect to appr.tc.
Capture audio frames to a WAV file in server example.
Show datachannel open / close events in server example.
Nothing published for this version
Nothing published for this version
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