AyCalls

Play audio and video into Telegram group voice chats from Python. No py-tgcalls. No tgcalls. No Telethon. Just Kurigram + aiortc + FFmpeg.

AyCalls is a small, opinionated library that wires three battle-tested tools together:

• Kurigram — MTProto signaling (phone.joinGroupCall, phone.LeaveGroupCall, …)
• aiortc — ICE / DTLS-SRTP / RTP, controlled at the ORTC level so SSRCs can be pinned
• FFmpeg — decodes anything you point it at into 48 kHz stereo PCM

The result is a pytgcalls-style API that just works: one play() call handles join, queue, playback, and auto-leave.

from aytgcalls import AyClient, AyCreds

client = AyClient(AyCreds.from_env())
await client.start()

# 🎵 Play local files, direct YouTube search, JioSaavn or live stream URLs
await client.play(-1001234567890, "song.mp3")                       # local audio
await client.play(-1001234567890, "yt:Alone Alan Walker")          # Instant YouTube search via aytube
await client.play(-1001234567890, "saavn:Kesariya")                # JioSaavn 320kbps via SaavnAPI
await client.play(-1001234567890, "https://youtu.be/1-xGerv5FOk")  # Direct YouTube URL

await client.pause(-1001234567890)
await client.resume(-1001234567890)
await client.skip(-1001234567890)
await client.end(-1001234567890)
await client.stop()

Every action takes chat_id first — one client, many voice chats.

---

Table of contents

• Features
• How it works
• Why you need two accounts
• Installation
• Generating a STRING_SESSION
• YouTube & JioSaavn Streaming
• Quick start
• Three usage styles
  • AyClient — one class, everything
  • AyFac — factory for many chats
  • AyCall — one call, full control
• API reference
• The queue
• Events
• Video / screen sharing
• Bot command surface
• Configuration
• Error handling
• Verifying it works
• Troubleshooting
• What's implemented, what's not
• Development
• License

---

Features

• 🚀 Native aytube YouTube Engine — Lightning fast, pure-Python stream extractor and search resolver with instant audio playback (yt:<query> or direct YouTube URLs). Zero yt-dlp overhead!
• 🎶 Native SaavnAPI (JioSaavn) Engine — Direct lossless 320kbps stream extraction with saavn:<query> or JioSaavn URLs.
• 🎵 Stream any audio format FFmpeg understands — MP3, FLAC, OGG/Opus, WAV, AAC, M4A, MOV, MKV, remote HTTP(S) streams, Telegram voice notes, radio feeds.
• 🎬 Video & screen sharing via crisp H.264 on presentation SSRC.
• ⏯ Full playback control — play, pause, resume, skip, previous, replay, stop, seek, forward, rewind, loop (track / queue / shuffle / count).
• 📜 Near-gapless transitions — the next track starts the moment the previous one hits EOF, maintaining the strict 20 ms WebRTC cadence.
• 🧠 Smart seeking & volume — byte-offset seeks for self-framing formats, FFmpeg -ss for header-dependent ones, digital software volume control (0%–200%).
• 🪄 One-call automation — play() auto-joins, auto-queues, and auto-leaves when the queue empties.
• 🔁 Auto-reconnect with exponential backoff on SFU disconnects.
• 🔇 Server-side controls — mute, unmute, volume through Telegram MTProto signaling.
• 👥 Participant inspection — query live WebRTC RTP / ICE / DTLS stats.
• 🪶 Zero bloat & pure Python WebRTC — built on modern aiortc and kurigram, completely eliminating py-tgcalls segfaults and C++ compilation nightmares.

How it works

chat_id
   │  channels.getFullChannel / messages.getFullChat
   ▼
InputGroupCall ──► phone.joinGroupCall(params = {ssrc, ufrag, pwd, fingerprints})
                              │
                              ▼
                   UpdateGroupCallConnection
                   {transport:{…}, audio:{payload-types, rtp-hdrexts}}
                              │
   ┌──────────────────────────┴───────────────────────────────────────┐
   │  ICE (controlling)  →  DTLS (client)  →  SRTP keys              │
   └──────────────────────────┬───────────────────────────────────────┘
                              ▼
 file/URL ─ FFmpeg ─► PCM s16le 48k/2 ─► ring buffer ─► gain ─► Opus 20 ms ─► RTP ─► SFU

The full protocol write-up — including how the SSRC and Opus payload type are pinned, and why we can't use RTCPeerConnection directly — lives in PROTOCOL.md.

Why you need two accounts

This is a Telegram limitation, not a library one: phone.joinGroupCall is user-only.

The standard pattern is a two-account setup:

Account	Role
User ("assistant"), via API_ID + API_HASH + STRING_SESSION	Joins the call and streams media. Required.
Bot, via BOT_TOKEN	Optional. Parses /play, /skip, etc. and dispatches to the assistant. Never touches the call.

GroupCall checks this at join time and raises BotClientNotAllowed if you hand it a bot session.

Installation

pip install aytgcalls

Remove conflicting Pyrogram forks

Kurigram is a maintained Pyrogram fork and still imports as pyrogram. If the real pyrogram or pyrofork is installed alongside it, the two packages overwrite each other's files and you get bizarre import errors. Clean them up first:

pip uninstall -y pyrogram pyrofork tgcalls py-tgcalls pytgcalls
pip install -U kurigram aytgcalls

Recommended installation (with ultra-fast engines + tgcrypto):

pip install "aytgcalls[all]"     # installs tgcrypto + aytube + SaavnAPI

Sanity-check the install:

python -c "import pyrogram; print(pyrogram.__version__)"   # should be a kurigram version
python -c "import importlib.util; assert importlib.util.find_spec('tgcalls') is None; print('no py-tgcalls')"

FFmpeg and libopus

# Debian / Ubuntu
sudo apt update && sudo apt install -y ffmpeg libopus0

# Fedora / RHEL
sudo dnf install -y ffmpeg opus

# macOS
brew install ffmpeg opus

Verify:

ffmpeg -version
python -c "from aytgcalls.media.opus import opus_available; print('opus ✓' if opus_available() else 'opus ✗')"

If ffmpeg isn't on PATH, point at it explicitly:

export AYTGCALLS_FFMPEG=/usr/local/bin/ffmpeg

Optional: YouTube / streaming sites

For YouTube, SoundCloud, Vimeo, Twitch and similar, install yt-dlp and the helper package picks it up automatically:

pip install -U yt-dlp

AyCalls uses YouTubeMusic internally — it wraps yt-dlp with format selectors that always return direct progressive URLs, never HLS / DASH manifests.

Generating a STRING_SESSION

Get API_ID / API_HASH from https://my.telegram.org → API development tools, then:

export API_ID=1234567
export API_HASH=0123456789abcdef0123456789abcdef
python examples/scripts/gen_session.py

Log in with the phone number of a normal account (not a bot). The script prints the session string once.

Treat STRING_SESSION like a password. Anyone with it has full access to that

YouTube & JioSaavn Streaming

AyCalls features built-in high-performance music streaming engines with zero manual configuration required:

1. YouTube via aytube

Powered by the ultra-fast, pure-Python aytube library. It extracts direct GoogleVideo audio streams and performs searches in milliseconds without heavy binary overhead:

# Instant YouTube search & stream
await client.play(chat_id, "yt:Alone Alan Walker")
await client.play(chat_id, "youtube:Starboy The Weeknd")

# Direct YouTube URLs (standard or shortened)
await client.play(chat_id, "https://www.youtube.com/watch?v=1-xGerv5FOk")
await client.play(chat_id, "https://youtu.be/1-xGerv5FOk")

2. JioSaavn via SaavnAPI

Powered by SaavnAPI for studio-grade 320 kbps AAC/M4A streams:

# Direct JioSaavn search & play
await client.play(chat_id, "saavn:Kesariya")
await client.play(chat_id, "saavn:Tum Hi Ho")

# Direct JioSaavn song URLs
await client.play(chat_id, "https://www.jiosaavn.com/song/tum-hi-ho/BgwicBdfAQU")

---

Quick start

import asyncio
from aytgcalls import AyClient, AyCreds

async def main():
    client = AyClient(AyCreds.from_env())
    await client.start()

    await client.play(-1001234567890, "song.mp3")    # auto-join + play
    await asyncio.sleep(30)
    await client.stop()

asyncio.run(main())

That's it — one play() call handles join, queue, play, and auto-leave.

Three usage styles

AyClient — one class, everything

The recommended entry point. One instance per bot, multiple voice chats:

from aytgcalls import AyClient, AyCreds

client = AyClient(AyCreds.from_env())
await client.start()

# every method takes chat_id first
await client.play(chat_id, "song.mp3")          # auto-join + play or queue
await client.pause(chat_id)
await client.resume(chat_id)
await client.skip(chat_id)
await client.seek(chat_id, 45)
await client.forward(chat_id, 10)
await client.rewind(chat_id, 10)
await client.volume(chat_id, 80)
await client.mute(chat_id)
await client.unmute(chat_id)
await client.loop(chat_id, "queue")
await client.shuffle(chat_id)
await client.play_video(chat_id, "clip.mp4")

# introspection
print(client.position(chat_id))
print(client.now_playing(chat_id))
print(client.is_connected(chat_id))
print(await client.get_participants(chat_id))   # who's in the call

# lifecycle
await client.end(chat_id)        # stop + leave this chat
await client.stop()              # leave all + shutdown

AyClient wraps AyFac internally and mirrors every per-chat control. You can also pass in a pre-built Kurigram Client if you need more control over startup.

AyFac — factory for many chats

For when you want the factory directly — useful if you already have a Kurigram client lying around:

from aytgcalls import AyFac

fac = AyFac(user_client)
await fac.play(chat_id, "song.mp3")         # creates, joins, plays or queues
await fac.pause(chat_id)
await fac.skip(chat_id)
await fac.volume(chat_id, 80)
await fac.stop(chat_id)                     # stop + leave
await fac.leave_all()                       # shutdown

AyCall — one call, full control

For a single persistent voice chat where you want the call object as the API:

from aytgcalls import AyCall

call = AyCall(user_client, chat_id)
await call.play("song.mp3")                 # joins + plays
await call.pause()
await call.resume()
await call.skip()
await call.end()                            # stop + leave

API reference

All methods are available on AyClient, AyFac (with chat_id first), and AyCall (without chat_id). The table below shows the AyClient signature.

Playback

await client.play(chat_id, source)              # join / play / queue, all automatic
await client.play(chat_id, source, force=True)  # jump the queue
await client.add(chat_id, source)               # alias for play()
await client.pause(chat_id)
await client.resume(chat_id)
await client.stop_playback(chat_id)             # stop audio, stay in the call
await client.stop(chat_id)                      # stop + leave (same as end)
await client.end(chat_id)
await client.previous(chat_id)
await client.replay(chat_id)

source can be a local path, an http(s) URL, a Kurigram Message, or an AudioSource.

Seeking

await client.seek(chat_id, 90)        # absolute seconds → returns where we landed
await client.forward(chat_id, 10)     # skip forward (default 10s)
await client.rewind(chat_id, 10)      # skip backward (default 10s, clamped at start)

Queue

await client.shuffle(chat_id)
await client.clear_queue(chat_id)
await client.remove(chat_id, 2)       # drop the track at index 2
await client.move(chat_id, 4, 0)      # move index 4 to the front
await client.loop(chat_id, "off")     # "off" | "track" | "queue" | "shuffle"
await client.loop(chat_id, 3)         # repeat current track 3 more times
await client.loop(chat_id)            # read current mode

Volume

await client.volume(chat_id, 80)      # local gain, 0..200 (%)
await client.mute(chat_id)            # server-side
await client.unmute(chat_id)          # server-side

Video

await client.play_video(chat_id, "clip.mp4")   # stream video
await client.stop_video(chat_id)               # stop video, audio continues

Introspection

client.now_playing(chat_id)           # TrackInfo: title, state, position, duration…
client.position(chat_id)              # current playback position (seconds)
client.duration(chat_id)              # track duration (None for live)
client.volume(chat_id)                # current volume setting
client.playback_state(chat_id)        # PlaybackState: PLAYING | PAUSED | IDLE…
client.is_connected(chat_id)          # whether we're in the voice chat
client.get_call(chat_id)              # underlying GroupCall, or None
await client.get_stats(chat_id)       # CallStats: packets, bytes, frames, ICE state…
await client.get_participants(chat_id)# list of participants
client.active_calls                   # dict of all joined calls
len(client)                           # number of active calls

Join / leave

await client.join(chat_id)            # join without playing
await client.leave(chat_id)           # leave this chat
await client.set_title(chat_id, "Late night 🎧")   # rename the voice chat

The queue

from aytgcalls import AyLoop

await call.queue.add("a.mp3")
await call.queue.add("b.mp3", position=0)         # insert at the front
await call.queue.extend(["c.mp3", "https://example.com/d.mp3"])
await call.queue.remove(1)
await call.queue.clear()
await call.queue.shuffle()
await call.queue.move(2, 0)

call.queue.current            # AudioSource | None
call.queue.items              # tuple of upcoming tracks
call.queue.history            # recently finished tracks
await call.queue.next()       # advance (respects loop mode)
await call.queue.previous()   # step back through history

await call.loop(3)                  # repeat the current track 3 more times
await call.loop("track")            # forever
await call.loop("queue")            # whole queue
await call.loop("shuffle")          # shuffle + keep looping
await call.loop("off")

loop() accepts a count, a LoopMode, or any friendly word (one, song, all, playlist, repeat, shuffle, off), so a chat command can be forwarded straight to it. After loop(3) plays the track three more times the mode falls back to off on its own.

play() returns (track, started_now) so a bot can reply either "playing" or "queued at #3" without tracking state itself. When a track ends the next one starts on its own.

Transitions are near-gapless: the next FFmpeg process starts the moment the previous one hits EOF, feeding the same ring buffer, so the 20 ms RTP cadence never breaks.

Events

@call.on_stream_end
async def _(call, source, reason):
    # reason: FINISHED | FAILED | STOPPED | TIMEOUT
    print("finished", source.display_name, reason.value)

@call.on_disconnect
async def _(call, reason):
    # reason: REQUESTED | CALL_ENDED | KICKED | TRANSPORT_FAILED | SFU_TIMEOUT | QUEUE_FINISHED
    print("disconnected", reason.value)

Video / screen sharing

await call.play_video("clip.mp4")       # stream a video file
await call.stop_video()                 # stop video, audio continues
await call.play("song.mp3")             # audio keeps going

The video track is encoded to H.264 by FFmpeg and sent on the dedicated presentation SSRC (ssrc-groups). Requires Telegram's presentation join path.

Bot command surface

See examples/bot_plus_assistant.py for a complete command set using AyFac:

/play <file|url>   or reply to a voice/audio message
/pause  /resume  /skip  /previous  /replay  /stop  /end
/seek <secs>  /forward [secs]  /rewind [secs]
/volume <0-200>  /mute  /unmute
/loop <n|track|queue|shuffle|off>
/remove <idx>  /move <from> <to>
/now   /queue   /participants

There is no /join and no /add — /play covers both.

Configuration

from aytgcalls import AyConfig

config = AyConfig(
    ffmpeg_path="ffmpeg",
    opus_bitrate=96_000,      # 64–128 kbps is Telegram's comfortable range
    buffer_ms=400,            # jitter buffer depth
    prefetch_ms=200,          # buffered before the first frame is released
    volume=100,
    ice_servers=(),           # Telegram's SFU is ICE-lite on a public IP; STUN not needed
    connect_timeout=20.0,
    keepalive_interval=10.0,  # phone.checkGroupCall
    auto_reconnect=True,
    reconnect_max_attempts=8,
)
call = AyCall(client, config=config)

Everything is also readable from the environment:

Variable	Meaning
API_ID, API_HASH, STRING_SESSION	user session (required)
BOT_TOKEN	optional command bot
AYTGCALLS_FFMPEG	path to the ffmpeg binary
AYTGCALLS_OPUS_BITRATE	Opus bitrate in bits/s
AYTGCALLS_BUFFER_MS, AYTGCALLS_PREFETCH_MS	buffering
AYTGCALLS_VOLUME	initial volume percent
AYTGCALLS_ICE_SERVERS	comma-separated STUN/TURN URLs
AYTGCALLS_CONNECT_TIMEOUT, AYTGCALLS_KEEPALIVE_INTERVAL	timeouts
AYTGCALLS_DEBUG=1	verbose logging with redacted signaling JSON

from aytgcalls import enable_debug
enable_debug()

Automation you can turn off

AyConfig(
    auto_join=True,          # play() joins by itself
    auto_leave=True,         # leave when the queue runs out
    auto_leave_delay=3.0,    # grace period, so a quick next request keeps the call
)

The grace period matters: if a user queues another song within auto_leave_delay, the pending leave is cancelled and the call stays up.

Error handling

AytgcallsError
├── BotClientNotAllowed          a bot session was used
├── GroupCallNotFound            no active voice chat in that chat
├── NotInGroup                   the account cannot access the chat
├── AlreadyJoined / NotJoined
├── AlreadyPlaying / NotPlaying
├── InvalidAudioSource / MediaSourceError
├── FFmpegError / FFmpegNotInstalled
├── OpusError
├── TransportError
│   ├── ICEFailed
│   └── DTLSHandshakeFailed
└── TelegramCallError            wraps DATA_JSON_INVALID, GROUPCALL_INVALID,
                                 GROUPCALL_FORBIDDEN, CHAT_ADMIN_REQUIRED,
                                 GROUPCALL_SSRC_DUPLICATE_MUCH, JOIN_AS_PEER_INVALID

TelegramCallError attaches a plain-English explanation to every known RPC error id:

from aytgcalls.exceptions import AytgcallsError

try:
    await client.play(chat_id, "song.mp3")
except AytgcallsError as exc:
    await message.reply(f"❌ {exc}")

Verifying it works on a VPS

export API_ID=... API_HASH=... STRING_SESSION=...
export TEST_CHAT_ID=-1001234567890         # a chat with a RUNNING voice chat
python examples/scripts/live_check.py

It joins, streams a tone, asserts that RTP packets actually left the host at the expected 50 packets/second, prints the stats, and leaves cleanly. Exit code 0 = pass.

✅ preflight: ffmpeg, libopus, no py-tgcalls
✅ user session: id=… @…
✅ joined. ssrc=1735203981
✅ ICE=completed  DTLS=connected
   packets_sent=601 (+600)  bytes=147840  frames=600 silence=3 underruns=3
✅ RTP is flowing at the expected 20 ms cadence
✅ real audio frames (not silence) reached the encoder
✅ left cleanly

Troubleshooting

Nobody can hear anything, but packets_sent keeps rising. The media path is fine; you're almost certainly server-side muted. In a group where members join muted, an admin must unmute the assistant, or the account needs speaking rights. Watch the logs for Server-side muted with can_self_unmute=False. You can also try await call.mute(False).

ICEFailed: ICE did not connect. Outbound UDP is blocked. Telegram's media servers need arbitrary outbound UDP (ports vary, commonly 40000–65535) to 91.108.x.x / 149.154.x.x. Cloud firewalls that only allow TCP will fail here. There's no TCP fallback in this package.

DTLSHandshakeFailed. ICE succeeded but the handshake didn't. Usually a stale call — leave, wait a few seconds, rediscover and rejoin (auto_reconnect=True does this for you). Also check your clock is correct — certificate validity is time-sensitive.

CHAT_ADMIN_REQUIRED. The account lacks rights for that action. Promote the assistant, or ask an admin to allow members to speak.

GROUPCALL_SSRC_DUPLICATE_MUCH. A previous session never left cleanly. AyCalls picks a fresh SSRC per join; wait a few seconds and retry.

FFmpegNotInstalled. apt install ffmpeg, or set AYTGCALLS_FFMPEG=/path/to/ffmpeg.

GroupCallNotFound. The voice chat isn't running (or is scheduled for later). AyCalls joins existing calls; it doesn't create them.

Choppy audio / lots of underruns in the stats. Increase buffer_ms / prefetch_ms, especially for remote URLs on a slow link.

ImportError mentioning pyrogram. Two forks are installed at once. See Remove conflicting Pyrogram forks.

TransportError: … RTMP/stream broadcast. Those calls don't accept WebRTC publishers at all — you'd have to push to the RTMP URL from phone.getGroupCallStreamRtmpUrl. See PROTOCOL.md §7.

What's implemented, what's not

Implemented:

• Discovery, join, keepalive, leave, mute/unmute, volume, update handling, reconnect
• One-call automation: play() auto-joins, auto-queues, the call auto-leaves when the queue empties
• Full playback control: play / pause / resume / skip / previous / replay / stop / end, seek / forward / rewind, loop (count / track / queue / shuffle), live position + duration
• Telegram voice notes and audio messages as sources (downloaded and cleaned up for you)
• JSON ⇄ SDP bridge (both directions, round-trip tested)
• ICE (controlling) + DTLS-SRTP (client) + RTP with pinned SSRC and the SFU's payload type
• FFmpeg → PCM → ring buffer → gain → Opus 20 ms → paced RTP
• Queue with loop / shuffle / history, near-gapless transitions, deterministic teardown
• Video / screen sharing via presentation SSRC (H.264, FFmpeg-encoded)
• Participant listing and voice-chat renaming

Deliberately not implemented:

• Receiving other participants' audio (we publish only)
• RTMP / stream-mode calls (detected and rejected with a clear error)
• E2E conference calls (public_key / block arguments of phone.joinGroupCall)

Development

git clone https://github.com/Sinchu-xD/AyCalls.git
cd AyCalls
pip install -e ".[dev]"
pytest -q          # 274 tests, no network required
ruff check .
mypy aytgcalls

Two of the test modules go further than unit tests:

• examples/tests/test_loopback.py stands up a second aiortc peer locally, serialises its parameters into exactly the JSON shape Telegram sends, and runs a real ICE + DTLS-SRTP + Opus RTP session against it.
• examples/tests/test_integration.py drives the public API — play() → pause / resume / skip / volume → leave() — against a fake Kurigram client that returns real TL objects and hands off to that local peer. It asserts the far end decodes 48 kHz audio, that phone.checkGroupCall keepalives run, that phone.leaveGroupCall gets our SSRC, and that no tasks or FFmpeg processes leak.

What neither can prove is that Telegram's production SFU accepts the join payload — that's what examples/scripts/live_check.py is for.

License

MIT — see LICENSE.