calm-optician-60209
02/15/2025, 8:20 AMgray-monkey-83525
02/15/2025, 8:20 AMcalm-optician-60209
02/15/2025, 9:28 AMcalm-optician-60209
02/16/2025, 9:53 AMgray-monkey-83525
02/16/2025, 9:53 AMagent.py
file serve different roles in your setup. The LiveKit Server acts as the core component of the LiveKit platform, handling signaling, media, and other real-time communication tasks. It is responsible for managing rooms, participants, and tracks, and can be deployed either on your infrastructure (self-hosted) or using LiveKit Cloud.
On the other hand, the agent.py
file is likely part of the LiveKit Agents framework, which is used to create programmable, server-side participants (agents) that can join LiveKit rooms. These agents can perform tasks such as speech-to-text, text-to-speech, and other AI-driven functionalities. The agent runs as a separate process and connects to the LiveKit Server to participate in rooms as an AI participant.
Would you like more detailed information on how to set up or use these components?
Sources:
- https://docs.livekit.io/agents/overview/ 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/17/2025, 4:28 PMgray-monkey-83525
02/17/2025, 4:28 PMVideoCodec
enum not having a defined value for 'H264'. This suggests that the LiveKit API or the specific version of the library you're using does not support 'H264' as a valid video codec option. This could be due to a mismatch between the expected codec values in your code and those defined in the library.
To resolve this, you might want to verify the available codec options in the LiveKit API documentation or the library version you are using. Ensure that 'H264' is indeed a supported codec, or consider using an alternative codec that is supported.
Would you like more detailed guidance on how to check the available codec options or update your code accordingly?
Sources:
- https://livekit-users.slack.com/archives/C07FRP66XGE/p1730288245.597009
- https://livekit-users.slack.com/archives/C048FRL1N2C/p1729871093.168429
Note: I wasn’t able to find highly relevant data sources, but above are a few potentially relevant links. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/17/2025, 4:30 PMgray-monkey-83525
02/17/2025, 4:30 PMVideoCodec
enum is being referenced in your code, specifically with the value 'H264'. Since you are only using audio, you should ensure that any video-related configurations are removed or set to None
if not needed. The error might be occurring because the code is attempting to set a video codec when it should be focusing solely on audio.
To resolve this, you can modify your start_azure_egress
function to exclude any video codec settings. Ensure that the RoomCompositeEgressRequest
and EncodedFileOutput
configurations are set for audio-only operations. Here's a simplified version of your code focusing on audio:
async def start_azure_egress(_ctx_: JobContext):
"""Start RoomComposite Egress to Azure Blob Storage"""
try:
# Initialize LiveKit API client
lkapi = api.LiveKitAPI(
os.getenv("LIVEKIT_HOST"),
os.getenv("LIVEKIT_API_KEY"),
os.getenv("LIVEKIT_API_SECRET")
)
# Azure Blob Storage configuration
azure_config = api.AzureBlobUpload(
account_name=os.getenv("AZURE_STORAGE_ACCOUNT"),
account_key=os.getenv("AZURE_STORAGE_KEY"),
container_name=os.getenv("AZURE_CONTAINER_NAME"),
)
# Generate unique filename with timestamp and phone number
phone_number = ctx.job.metadata.replace('+', '') if ctx.job.metadata else 'unknown'
filename = f"recordings/{datetime.utcnow().isoformat()}_{phone_number}.mp4"
# Configure file output with Azure settings
file_output = api.EncodedFileOutput(
file_type=api.EncodedFileType.MP4,
filepath=filename,
azure=azure_config,
audio_codec=api.AudioCodec.AAC,
# Remove or set video_codec to None
video_codec=None,
preset=None, # Remove video preset
)
# Start RoomComposite Egress with auto-start configuration
egress_request = api.RoomCompositeEgressRequest(
room_name=ctx.room.name,
layout="speaker-dark", # Audio-focused layout
audio_only=True, # Force audio-only recording
file_outputs=[file_output],
options={
"auto_start": True, # Enable auto-start
"wait_for_participants": True # Wait for participants before starting
}
)
egress_info = await lkapi.egress.start_room_composite_egress(egress_request)
logger.info(f"Egress started: {egress_info.egress_id}")
return egress_info
except Exception as e:
logger.error(f"Failed to start egress: {str(e)}")
logger.error(f"Stack trace: {traceback.format_exc()}")
raise
Would you like more detailed guidance on configuring audio-only egress?
Sources:
- Track Egress
- Composite and Web Recordings 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/17/2025, 4:33 PMgray-monkey-83525
02/17/2025, 4:33 PMlivekit_host
is being used, such as a configuration file or a specific part of your code. If this doesn't help, please reach out to the Livekit support team for further assistance. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/17/2025, 4:37 PMgray-monkey-83525
02/17/2025, 4:37 PMEncodedFileOutput
object not having an audio_codec
field. This suggests that the audio_codec
parameter is not a valid attribute for the EncodedFileOutput
class in the LiveKit API you are using.
To resolve this, you should remove the audio_codec
field from the EncodedFileOutput
configuration. Instead, focus on configuring the output settings that are supported by the EncodedFileOutput
class, such as file_type
, filepath
, and any storage-specific configurations like azure
.
Here's a simplified version of your code without the audio_codec
field:
async def start_azure_egress(_ctx_: JobContext):
"""Start RoomComposite Egress to Azure Blob Storage"""
try:
# Initialize LiveKit API client
lkapi = api.LiveKitAPI(
os.getenv("LIVEKIT_HOST"),
os.getenv("LIVEKIT_API_KEY"),
os.getenv("LIVEKIT_API_SECRET")
)
# Azure Blob Storage configuration
azure_config = api.AzureBlobUpload(
account_name=os.getenv("AZURE_STORAGE_ACCOUNT"),
account_key=os.getenv("AZURE_STORAGE_KEY"),
container_name=os.getenv("AZURE_CONTAINER_NAME"),
)
# Generate unique filename with timestamp and phone number
phone_number = ctx.job.metadata.replace('+', '') if ctx.job.metadata else 'unknown'
filename = f"recordings/{datetime.utcnow().isoformat()}_{phone_number}.mp4"
# Configure file output with Azure settings
file_output = api.EncodedFileOutput(
file_type=api.EncodedFileType.MP4,
filepath=filename,
azure=azure_config
)
# Start RoomComposite Egress with auto-start configuration
egress_request = api.RoomCompositeEgressRequest(
room_name=ctx.room.name,
layout="speaker-dark", # Audio-focused layout
audio_only=True, # Force audio-only recording
file_outputs=[file_output],
options={
"auto_start": True, # Enable auto-start
"wait_for_participants": True # Wait for participants before starting
}
)
egress_info = await lkapi.egress.start_room_composite_egress(egress_request)
logger.info(f"Egress started: {egress_info.egress_id}")
return egress_info
except Exception as e:
logger.error(f"Failed to start egress: {str(e)}")
logger.error(f"Stack trace: {traceback.format_exc()}")
raise
Would you like more detailed guidance on configuring egress for audio-only recordings?
Sources:
- https://docs.livekit.io/home/egress/api/
- https://docs.livekit.io/home/egress/room-composite/
Note: I wasn’t able to find highly relevant data sources, but above are a few potentially relevant links. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/18/2025, 6:02 PMgray-monkey-83525
02/18/2025, 6:02 PMdisconnect
method on the room object. This will remove the agent from the session.
2. Mute the Agent: If you prefer to keep the agent in the room but mute it, you can use the mute_published_track
method to mute the agent's audio track.
Here's a basic example of how you might implement these actions:
async def handle_warm_transfer(ctx: JobContext):
# Option 1: Disconnect the agent
await ctx.room.disconnect()
logger.info("Agent disconnected from the room.")
# Option 2: Mute the agent
# Assuming you have access to the agent's track publication
# await ctx.room.mute_published_track(track_sid, muted=True)
# logger.info("Agent's audio track muted.")
Would you like more detailed guidance on implementing these actions in your specific setup?
Sources:
- Managing Participants 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/18/2025, 6:09 PMgray-monkey-83525
02/18/2025, 6:09 PMdisconnect
method on the room object. This will remove the agent from the session entirely. Here's how you can modify your warm_transfer_with_fallback
function to achieve this:
@llm.ai_callable()
async def warm_transfer_with_fallback(self):
"""Initiate warm transfer to human agent with 20s timeout and fallback contact collection"""
try:
agent_number = "<tel:+16147453|+161473>"
# Get local participant with null check
local_participant = self.room.local_participant
if not local_participant:
logger.error("No local participant found")
return "TRANSFER_FAILED"
# Announce transfer using TTS
await self.agent.say("Connecting you to a specialist...")
# Create SIP participant with initial DTMF
sip_response = await self.api.sip.create_sip_participant(
api.CreateSIPParticipantRequest(
room_name=self.room.name,
sip_trunk_id=os.getenv("SIP_OUTBOUND_TRUNK_ID"),
sip_call_to=agent_number,
participant_identity=f"agent_{agent_number.replace('+', '')}",
participant_name="Human Agent",
play_dialtone=True,
dtmf="ww0" # Initial DTMF sequence
)
)
# Wait for human agent
human_joined = await asyncio.wait_for(
self._wait_for_human_agent(agent_number),
timeout=20
)
if not human_joined:
raise asyncio.TimeoutError("Human agent did not join in time")
# Disconnect the agent from the room
await self.room.disconnect()
logger.info("Agent disconnected from the room.")
return "TRANSFER_COMPLETE"
except (asyncio.TimeoutError, Exception) as e:
logger.error(f"Transfer failed: {str(e)}")
await self.agent.say(
"I'm sorry, we're having trouble connecting you. "
"Please provide your contact details so we can reach you."
)
return "TRANSFER_FAILED"
except Exception as e:
logger.error(f"Transfer error: {e}")
return "TRANSFER_FAILED"
This code will disconnect the agent from the room once the SIP participant is created and the human agent joins. If the transfer fails, it will inform the user and not attempt to restore the agent's audio state since the agent is disconnected.
Would you like more detailed guidance on implementing this in your setup?
Sources:
- Room.disconnect 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/18/2025, 6:14 PMcalm-optician-60209
02/18/2025, 6:14 PMgray-monkey-83525
02/18/2025, 6:14 PMLIVEKIT_HOST
, LIVEKIT_API_KEY
, and LIVEKIT_API_SECRET
.
2. Create the SIP Participant: Use the create_sip_participant
method to initiate a call to the human agent. This involves specifying the room name, SIP trunk ID, and the target phone number.
3. Wait for the Human Agent: Implement a mechanism to wait for the human agent to join the room. This can be done using a loop that checks for the presence of the human agent's participant ID in the room's remote participants.
4. Disconnect the Agent: Once the human agent joins, call the disconnect
method on the room object to remove the agent from the session. This ensures that the agent is no longer part of the conversation.
5. Handle Transfer Failures: Implement error handling to manage scenarios where the transfer fails, such as timeouts or exceptions. Inform the user and provide a fallback mechanism, like collecting contact details.
Here's a detailed implementation:gray-monkey-83525
02/18/2025, 6:15 PMimport asyncio
import os
from livekit import api, rtc
from livekit.agents import JobContext
import logging
logger = logging.getLogger(__name__)
async def warm_transfer_with_fallback(ctx: JobContext):
"""Initiate warm transfer to human agent with 20s timeout and fallback contact collection"""
try:
agent_number = "<tel:+16147453|+161473>"
# Get local participant with null check
local_participant = ctx.room.local_participant
if not local_participant:
logger.error("No local participant found")
return "TRANSFER_FAILED"
# Announce transfer using TTS
await ctx.agent.say("Connecting you to a specialist...")
# Create SIP participant with initial DTMF
sip_response = await ctx.api.sip.create_sip_participant(
api.CreateSIPParticipantRequest(
room_name=ctx.room.name,
sip_trunk_id=os.getenv("SIP_OUTBOUND_TRUNK_ID"),
sip_call_to=agent_number,
participant_identity=f"agent_{agent_number.replace('+', '')}",
participant_name="Human Agent",
play_dialtone=True,
dtmf="ww0" # Initial DTMF sequence
)
)
# Wait for human agent
human_joined = await asyncio.wait_for(
wait_for_human_agent(ctx, agent_number),
timeout=20
)
if not human_joined:
raise asyncio.TimeoutError("Human agent did not join in time")
# Disconnect the agent from the room
await ctx.room.disconnect()
<http://logger.info|logger.info>("Agent disconnected from the room.")
return "TRANSFER_COMPLETE"
except (asyncio.TimeoutError, Exception) as e:
logger.error(f"Transfer failed: {str(e)}")
await ctx.agent.say(
"I'm sorry, we're having trouble connecting you. "
"Please provide your contact details so we can reach you."
)
return "TRANSFER_FAILED"
async def wait_for_human_agent(ctx: JobContext, agent_number: str) -> bool:
"""Monitor room for human agent connection"""
try:
human_agent_id = f"agent_{agent_number.replace('+', '')}"
for _ in range(40): # Check every 0.5s for 20s total
if ctx.room.remote_participants and human_agent_id in ctx.room.remote_participants:
return True
await asyncio.sleep(0.5)
return False
except Exception as e:
logger.error(f"Error waiting for human agent: {e}")
return False
gray-monkey-83525
02/18/2025, 6:15 PMdisconnect
method to remove the agent from the room once the transfer is successful.
This implementation provides a structured approach to handling warm transfers with fallback mechanisms in LiveKit. If you have any specific questions or need further customization, feel free to ask!
Sources:
- https://livekit-users.slack.com/archives/C07FRP66XGE/p1729561674.416969
- https://livekit-users.slack.com/archives/C07FY8WHGPM/p1728095142.517319 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.calm-optician-60209
02/18/2025, 7:06 PMgray-monkey-83525
02/18/2025, 7:06 PMAPITimeoutError
suggests that the request to OpenAI's API is taking too long to complete, resulting in a timeout.
Possible Causes:
1. Network Latency: The connection to OpenAI's API might be experiencing high latency, causing the request to exceed the timeout limit.
2. API Rate Limiting: If you're making too many requests in a short period, you might be hitting rate limits, which can lead to delays and timeouts.
3. Server Load: The OpenAI servers might be under heavy load, causing delays in processing requests.
Suggested Solutions:
- Increase Timeout: If possible, increase the timeout setting for the API request to allow more time for the response.
- Retry Logic: Implement a retry mechanism with exponential backoff to handle transient network issues or server load.
- Check Network: Ensure that your network connection is stable and has low latency to the OpenAI servers.
- Monitor API Usage: Check your API usage to ensure you are not hitting rate limits.
Here's a basic example of how you might implement a retry mechanism:
import asyncio
import openai
async def generate_completion_with_retry(prompt, max_retries=3):
attempt = 0
while attempt < max_retries:
try:
response = await openai.Completion.create(
engine="davinci",
prompt=prompt,
max_tokens=150
)
return response
except openai.error.Timeout as e:
attempt += 1
if attempt >= max_retries:
raise e
await asyncio.sleep(2 ** attempt) # Exponential backoff
# Usage
try:
result = await generate_completion_with_retry("Hello, world!")
print(result)
except openai.error.Timeout:
print("Failed to generate completion after multiple attempts.")
Would you like more detailed guidance on implementing these solutions?
Sources:
- https://livekit-users.slack.com/archives/C07FY8WHGPM/p1732345136.594869
- https://livekit-users.slack.com/archives/C07FY8WHGPM/p1732682471.692099 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.