SIP vs WebRTC: Voice agent latency and audio trade-offs

The latency breakdown: Transport vs model execution

Every voice conversation with an automated agent incurs latencies from multiple stages. The total turnaround time includes speech-to-text processing, language model inference, text-to-speech generation, and network transport. While engineers focus heavily on model speed, transport choice sets the baseline floor for real-time responsiveness.

Voice agents operate over two primary transport protocols: Session Initiation Protocol (SIP) and Web Real-Time Communication (WebRTC). SIP connects to traditional public switched telephone networks (PSTN) via carriers like Twilio, RingCentral, and Vonage. WebRTC creates direct browser connections using standard web protocols. Choosing between them determines whether your transport overhead is 50 milliseconds or 300 milliseconds before your models process a single byte of audio.

SIP trunking: Phone system compatibility at the cost of packet hops

SIP trunking remains the standard for phone-based customer interactions. When a customer dials a phone number from a mobile phone or landline, the call routes through carrier switches, hits a SIP trunk provider, and arrives at your voice server. This path introduces network hops and audio transcoding steps that add non-negotiable delay.

Standard phone calls use narrow-band audio codecs, usually G.711 sampled at 8 kHz. Converting this audio between PSTN networks and digital media servers creates jitter buffer delays. On average, network transport over SIP trunking adds 150 to 350 milliseconds of latency to a round-trip conversation turn.

Despite this delay, SIP is necessary for traditional inbound phone coverage. Most customers dial phone numbers directly rather than opening a web page. To compensate for PSTN transport lag, specialized teams focus on optimizing the backend orchestration layer. Managed agent services like AutoAppoint deploy inbound voice agents that pick up calls in under two seconds 24/7, absorbing PSTN transit delays by optimizing model execution and integration routing across tools like HubSpot, Salesforce, or GoHighLevel.

WebRTC: Sub-100ms transport for browser channels

WebRTC handles real-time media directly within web browsers and mobile applications. It establishes a direct media connection between the user interface and the backend server, bypassing carrier networks and telephony gateways entirely.

By eliminating PSTN routing, WebRTC reduces network transport latency to 30 to 100 milliseconds. Audio streams move directly from the user's microphone to the processing pipeline. WebRTC also negotiates media parameters dynamically, adapting to network jitter without adding fixed buffer delays. When latency reduction is the primary objective, WebRTC outperforms SIP trunking across every transport metric.

However, WebRTC requires users to remain inside a web browser or mobile application with an active internet connection. It cannot receive a traditional inbound telephone call from a landline. If your primary lead channel is a published phone number, WebRTC cannot replace your SIP provider.

Codec choice and speech-to-text accuracy

Transport protocol directly dictates audio clarity. Audio fidelity impacts how quickly and accurately speech-to-text models parse spoken words.

SIP trunking forces audio through G.711 or wide-band G.722 codecs. These compress audio frequencies above 3.4 kHz or 7 kHz. Background noise, heavy accents, and rapid speech patterns suffer degradation over G.711 streams. When speech-to-text engines receive low-bitrate audio, word error rates increase. Higher error rates force language models to process ambiguous input, adding downstream inference cycles.

WebRTC utilizes the Opus codec by default. Opus samples full-band audio at 48 kHz, preserving high-frequency consonants and vocal nuances. Higher audio fidelity yields clean transcriptions on the first pass. This reduces the need for secondary context checks in language models. Proficiency Workflow's coverage of real-time voice latencies highlights how codec selection directly influences total response delay and webhook stability during high-volume spikes.

Architecting for inbound calls and web workflows

Selecting the right transport layer depends on where customer conversations originate. Teams building browser widgets or app-embedded voice assistants should use WebRTC. It delivers low latency, full-band audio, and direct client-side metadata passing.

Businesses handling inbound phone traffic must rely on SIP trunking. To maintain acceptable response times over PSTN connections, focus on optimizing your backend infrastructure:

• Stream audio directly: Use WebSocket connections from your SIP provider to stream raw PCM audio straight into speech recognition engines without file-based buffers.
• Co-locate servers: Place voice orchestration servers in the same cloud region as your SIP provider's media gateways to eliminate geographical packet transit time.
• Manage state asynchronously: Decouple calendar lookups and CRM updates from the live audio loop. Use asynchronous webhooks to query tools like Pipedrive, Zoho, or Google Calendar without stalling speaker turns.

For additional details on managing concurrent calls and system load, review our previous report on voice agent dispatch and concurrency benchmarks. Matching the right transport protocol to your channel constraints ensures fast, reliable voice interactions for every caller.