LiveKit integration combines real-time communication capabilities with traditional telephony infrastructure. By connecting LiveKit to SIP trunks and PSTN networks, companies can enable users to make and receive calls, use standard phone numbers, and route voice traffic through SIP providers and telecom carriers. Clover Dynamics handles SIP trunk configuration, call routing, DID number setup, connection parameters, PBX integration, and scalable HD voice workflows.
LiveKit SIP PSTN Integration
Discover how you can ensure reliable inbound and outbound communication, call routing, and support for AI voice agents using LiveKit Server solutions. Securely and seamlessly integrate LiveKit with SIP trunks, PSTN networks, and existing telephony systems with Clover Dynamics.
Connect real-time audio and voice applications to traditional telephony infrastructure. LiveKit SIP PSTN integration involves configuring SIP connectivity, carrier integrations, and call routing, enabling companies to connect LiveKit with the PSTN, SIP providers, and existing telephony systems, empowering real-time WebRTC applications and AI voice agents.
Our LiveKit SIP PSTN Integration Services
- 01
SIP Trunk Integration
Enable voice traffic transmission between real-time applications and telephone networks using [LiveKit integrations](https://www.cloverdynamics.com/expertise/technologies/livekit-integrations). We handle SIP trunk configuration, endpoints, seamless call routing, and connection parameters to ensure stable two-way communication.
- 02
PSTN Connectivity
Let your users make and accept calls in real time. Our team helps configure numbers, routing, and connection parameters to ensure reliable connectivity with mobile and landline networks by connecting LiveKit to the PSTN.
- 03
Inbound and Outbound Calling
Configure outbound and inbound calls using LiveKit SIP, including DID number handling, routing, and call transfers between SIP participants. This integration enables applications to receive phone calls and initiate outbound calls, connecting users with voice agents, operators, and real-time communication services.
- 04
LiveKit SIP Configuration
Connect your telephony infrastructure with real-time applications. We configure SIP participants, trunks, numbers, routing rules, and call parameters tailored to your architecture, ensuring stable audio transmission and predictable SIP call behavior.
- 05
Carrier Integration
Integrate LiveKit with SIP carriers and telephony providers, including trunk connections, phone numbers, and routing rules. We tailor the integration to provider requirements to ensure reliable voice traffic exchange and scalable PSTN connectivity.
- 06
Existing Telephony Integration
We integrate SIP servers, PBX systems, telephony platforms, and other components while preserving current call-handling processes, enabling companies to gradually introduce real-time communication, voice AI agents, and modern voice workflows.
Inbound and Outbound Calling With LiveKit SIP
Inbound PSTN Calls
Configure incoming PSTN calls via LiveKit SIP by linking phone numbers to voice applications, operators, or AI voice agents for real-time call handling.
Outbound Dialing
Implement outbound dialing via LiveKit SIP so applications can initiate calls to mobile and landline numbers through connected SIP trunks and carriers.
DID Numbers
Connect and configure DID numbers to receive incoming calls and identify specific lines, users, or services within the telephony infrastructure.
Call Routing
Automatically route calls to the appropriate SIP participants, voice agents, operators, or phone numbers based on defined rules.
SIP Participants
Connect SIP participants to LiveKit rooms so phone users can interact with other participants, AI agents, and real-time communication workflows.
Transfers and DTMF
Implement call transfers and DTMF, including transfers between agents, forwarding to other numbers, and processing touch-tone signals during a call.
Related Case Study
AI-Powered Customer Support Agent for Call Center
Inbound PSTN Calls
Configure incoming PSTN calls via LiveKit SIP by linking phone numbers to voice applications, operators, or AI voice agents for real-time call handling.
Outbound Dialing
Implement outbound dialing via LiveKit SIP so applications can initiate calls to mobile and landline numbers through connected SIP trunks and carriers.
DID Numbers
Connect and configure DID numbers to receive incoming calls and identify specific lines, users, or services within the telephony infrastructure.
Call Routing
Automatically route calls to the appropriate SIP participants, voice agents, operators, or phone numbers based on defined rules.
SIP Participants
Connect SIP participants to LiveKit rooms so phone users can interact with other participants, AI agents, and real-time communication workflows.
Transfers and DTMF
Implement call transfers and DTMF, including transfers between agents, forwarding to other numbers, and processing touch-tone signals during a call.
Connect LiveKit to SIP Trunks and PSTN Networks
SIP Servers and PBX Integrations
LiveKit Integration With Asterisk, FreeSWITCH, Kamailio, and OpenSIPS
Connect LiveKit to SIP servers and PBX platforms while preserving existing infrastructure. We configure SIP routing, registration, trunk connections, inbound and outbound calls, transfers, DTMF, and stable audio transmission between LiveKit and external systems.
Call Routing
Automatically route calls to the appropriate SIP participants, voice agents, operators, or phone numbers based on defined rules.
Transfers and DTMF
Implement call transfers and DTMF, including transfers between agents, forwarding to other numbers, and processing touch-tone signals during a call.
LiveKit Integration With Asterisk, FreeSWITCH, Kamailio, and OpenSIPS
Connect LiveKit to SIP servers and PBX platforms while preserving existing infrastructure. We configure SIP routing, registration, trunk connections, inbound and outbound calls, transfers, DTMF, and stable audio transmission between LiveKit and external systems.
Call Routing
Automatically route calls to the appropriate SIP participants, voice agents, operators, or phone numbers based on defined rules.
Transfers and DTMF
Implement call transfers and DTMF, including transfers between agents, forwarding to other numbers, and processing touch-tone signals during a call.
LiveKit SIP Carrier Integrations
LiveKit SIP Integration With Twilio, Telnyx, Vonage, and Other Carriers
We configure SIP trunks, DID numbers, inbound and outbound calling, routing, failover, AI voice agent connections, existing number usage, and audio-quality verification for leading telephony providers.
Related Case Study
AI-Powered Conversational Bot for Digital Entertainment
LiveKit SIP Integration With Twilio, Telnyx, Vonage, and Other Carriers
We configure SIP trunks, DID numbers, inbound and outbound calling, routing, failover, AI voice agent connections, existing number usage, and audio-quality verification for leading telephony providers.
LiveKit SIP for AI Voice Agents
AI Voice Agent Telephony Workflows
Use telephony channels as an interface for AI voice agents by integrating LLMs, speech-to-text, text-to-speech, RAG, MCP, and external APIs via LiveKit SIP. We build production-ready architectures with conversation context, tool calling, turn detection, human handoff, call recording, and integrations with CRMs, booking systems, and internal services.
Existing Telephony Integration
We integrate SIP servers, PBX systems, telephony platforms, and other components while preserving current call-handling processes, enabling companies to gradually introduce real-time communication, voice AI agents, and modern voice workflows.
AI Voice Agent Telephony Workflows
Use telephony channels as an interface for AI voice agents by integrating LLMs, speech-to-text, text-to-speech, RAG, MCP, and external APIs via LiveKit SIP. We build production-ready architectures with conversation context, tool calling, turn detection, human handoff, call recording, and integrations with CRMs, booking systems, and internal services.
Existing Telephony Integration
We integrate SIP servers, PBX systems, telephony platforms, and other components while preserving current call-handling processes, enabling companies to gradually introduce real-time communication, voice AI agents, and modern voice workflows.
Build AI-Native systems with intelligence at the core - Not as an add-on
LiveKit SIP Solutions and Use Cases We Build
Low-Latency Video and Audio
Stable real-time interaction that adapts when network quality changes.
Screen Sharing
Controlled sharing of screens and individual application windows.
Participant Management
Connection, state, media stream, and participant action management.
Role-Based Permissions
Rules for speaking, screen sharing, recording, and moderation.
Real-Time Messaging
Low-latency text messages and events inside a session.
Recording
Audio and video recording for review, storage, analytics, and workflows.
Live Transcription
Real-time speech-to-text for subtitles, notes, search, and processing.
AI Voice Assistants
Conversational assistants that respond to requests and perform actions.
Virtual Backgrounds
Replace or obscure the user’s physical background during video calls.
Adaptive Streaming
Adjust media quality to bandwidth and network conditions.
Notifications
Incoming call, participant, session-change, and custom event notifications.
Analytics
Metrics for sessions, participants, quality, feature usage, and performance.
Authentication
Secure access, room controls, and end-to-end protection for media resources.
Moderation
Blocking, muting, access management, and rule-enforcement tools.
Session Management
Room lifecycle, participant connections, statuses, and access parameters.
Related Case Study
AI-Integrated Web Platform for Engaging Movie & TV Discussions
Low-Latency Video and Audio
Stable real-time interaction that adapts when network quality changes.
Screen Sharing
Controlled sharing of screens and individual application windows.
Participant Management
Connection, state, media stream, and participant action management.
Role-Based Permissions
Rules for speaking, screen sharing, recording, and moderation.
Real-Time Messaging
Low-latency text messages and events inside a session.
Recording
Audio and video recording for review, storage, analytics, and workflows.
Live Transcription
Real-time speech-to-text for subtitles, notes, search, and processing.
AI Voice Assistants
Conversational assistants that respond to requests and perform actions.
Virtual Backgrounds
Replace or obscure the user’s physical background during video calls.
Adaptive Streaming
Adjust media quality to bandwidth and network conditions.
Notifications
Incoming call, participant, session-change, and custom event notifications.
Analytics
Metrics for sessions, participants, quality, feature usage, and performance.
Authentication
Secure access, room controls, and end-to-end protection for media resources.
Moderation
Blocking, muting, access management, and rule-enforcement tools.
Session Management
Room lifecycle, participant connections, statuses, and access parameters.
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Secure Messaging App Development for Business
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What you get with Clover
We’ve helped startups raise over $72M in follow-on funding after launching products built by our teams
Our Awards
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Clover Dynamics has been key to our rapid growth. They delivered exceptional quality and complex features, helping us achieve triple-digit growth in under 5 months. I highly advise any startup looking to find the right tech capabilities to collaborate with Clover Dynamics.
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LiveKit SIP PSTN Integration FAQs
It connects real-time applications to the public telephone network through SIP, enabling LiveKit to handle incoming and outgoing phone calls.
LiveKit's SIP service connects through a SIP trunk provider, using inbound and outbound trunks to receive and make calls.
Yes. LiveKit supports incoming and outgoing calls through SIP trunks, phone numbers, SIP URIs, and configured dispatch rules.
LiveKit is compatible with most providers that support standard SIP trunking and phone-network connectivity.
Yes. The integration is implemented through SIP trunks while preserving existing telephony infrastructure and call routing.
Yes. LiveKit SIP can connect AI agents to phone calls and use AI models, agent workers, and real-time audio to automate voice workflows.
Yes. LiveKit can route calls between participants, process tone signals, and manage transfers and telephony workflows.
We use secure trunking, protected SIP connections, access control, and proper configuration of trunk credentials, caller identity, and call routing.