Voice AI That Speaks Carrier-Grade SIP

Session Timers, Re-INVITEs, QoS Preconditions. The SIP features every carrier network depends on, built by the team that wrote the SIP stack the industry runs on.

20+ years of production SIP engineering

< 1.2s typical AI response latency

2,000+ companies in production

2.7B+ minutes processed on the platform

Why WebRTC Platforms Break on Real Carriers

Session Timers are mandatory in carrier networks

Periodic SIP re-INVITE or UPDATE messages keep a call session alive. Without them, long-running calls die silently when a network element times out.

QoS Preconditions guarantee voice quality

IMS networks negotiate quality-of-service parameters before the media session starts. A platform built for WebRTC rooms cannot participate in this negotiation.

Re-INVITE handling is basic telephony

Hold, resume, codec renegotiation, adding or removing media streams, and session refresh all happen through Re-INVITEs.

SIP dialog state gets lost in translation

When a SIP call becomes a participant in a room, headers, session parameters, and authentication context get reduced to a room participant object.

Build a Voice AI Agent

Supported Languages

• Python
• TypeScript
• Go
• Java
• Ruby
• PHP
• Perl
• C++
• C#

Example Code

from signalwire import AgentBase
from signalwire.core.function_result import FunctionResult

class SupportAgent(AgentBase):
    def __init__(self):
        super().__init__(name="Support Agent", route="/support")
        self.prompt_add_section("Instructions",
            body="You are a customer support agent. "+
            "Greet the caller and resolve their issue.")
        self.add_language("English", "en-US", "rime.spore:mistv2")

@AgentBase.tool(name="check_order")
    def check_order(self, order_id: str):
        """Check the status of a customer order.
        Args:
            order_id: The order ID to look up
        """
        return FunctionResult(f"Order {order_id}: shipped, ETA April 2nd")

agent = SupportAgent()
agent.run()

Room-Based Architecture vs. Telephony-Native

Room-Based Platform

• SIP features must map to room concepts, and most have no equivalent
• SIP is a bridge feature; carrier edge cases get secondary treatment
• Session Timers and QoS Preconditions are difficult to preserve
• Re-INVITE and dialog state are reduced to participant events
• Operational behavior changes when carrier traffic increases

SignalWire

• SIP and media processing share one telephony-native engine
• Session Timers, QoS Preconditions, and Re-INVITEs are handled natively
• AI runs inside the carrier-grade media stack
• The same behavior holds from one call to carrier scale
• Built by the team that created FreeSWITCH
• Full SIP dialog state remains available through the call lifecycle

Deploy Voice AI on Carrier Infrastructure

Define your agent

Write a YAML document or Python class. The same definition works whether the call arrives from an IMS carrier, SIP trunk, or PSTN line.

Connect your SIP infrastructure

Point SIP trunks at SignalWire. Session Timers, QoS Preconditions, and Re-INVITEs are handled transparently by the FreeSWITCH-based stack.

Test with real carrier traffic

Route test calls through your IMS or enterprise telephony environment and verify SIP compliance end-to-end.

Scale to production

Same platform, same SIP stack, same behavior at carrier scale. No surprises when traffic increases.

Sigmond Runs on SignalWire

A voice and video AI agent built on the SignalWire SDK, wired to a live knowledge base, able to see what you show him. Build the same thing on voice, phone, WhatsApp, or SIP, or an entire white-label platform with SignalWire as the network underneath.

Call recording

This call is recorded. Audio and transcripts are used to improve our AI and customer support experience, and aren't shared or sold to third parties.

Trusted by 2,000+ Companies

Built by the team that wrote the SIP stack.

Deploy voice AI on real carrier infrastructure with full SIP compatibility from day one.