WebRTC and SIP both carry live voice to an AI agent. Voice AI platforms list them side by side, so teams building an AI receptionist often expect to pick one or the other for connecting callers to the agent. In practice, each protocol serves a different way customers reach the business.
An AI receptionist usually has two front doors. The first is your business phone number: a customer calls from a mobile phone or landline, and SIP connects that call from the telephone network to your voice AI agent. The second is your website or app: a visitor clicks a “talk to us” button, and WebRTC streams audio from their browser microphone to the same agent. A receptionist that answers both doors needs both protocols.
So the real decision isn’t WebRTC vs SIP. It’s understanding what each protocol does, how they work together on a single call, and which doors your AI receptionist actually needs to open. This post walks through all three, plus the one scenario where you genuinely only need one. Let’s start with a clean picture of what each protocol does.
WebRTC vs SIP: What Each Protocol Actually Does
WebRTC and SIP are easy to confuse because both show up whenever “real-time voice” is mentioned. But they solve different problems. The cleanest way to keep them straight is to tie each one to a front door: SIP is the phone side, WebRTC is the screen side.
SIP: the phone side
SIP (Session Initiation Protocol) is a signaling protocol. Its job is call control: setting up a call, managing it, and tearing it down. It decides who called whom, when to answer, which codec to use, and when to hang up.
A common point of confusion: SIP doesn’t carry the audio itself. The voice travels separately over RTP (Real-time Transport Protocol). SIP is the call’s control channel; RTP is the media. Keeping that split in mind saves a lot of debugging later.
For a receptionist, SIP is what connects the agent to real phone numbers over the public switched telephone network (PSTN), by way of a carrier’s SIP trunk. That connection is what unlocks the phone-side jobs:
- Answering an inbound business line
- Covering calls after hours
- Placing outbound reminders or callbacks
- Replacing an aging IVR menu on an existing number (a DID)
- Warm-transferring a caller to a human (via SIP REFER — more on this shortly)
One detail to file away: the moment a call touches the PSTN, its audio is typically carried in G.711, a narrowband codec sampled at 8 kHz. That constraint comes back to bite in a way most teams don’t expect, which we’ll get to.
WebRTC: the screen side
WebRTC (Web Real-Time Communication) is a browser-native suite for real-time audio, video, and data. No plugins, no installs. A web page can capture a microphone and start streaming the moment it loads. It runs with sub-second latency, is encrypted by default (DTLS-SRTP), and uses the Opus codec, which adapts its quality to network conditions.
For a receptionist, WebRTC owns everything that starts on a screen:
- The “talk to us” widget embedded on your website
- An in-app voice assistant on mobile
- A browser-based softphone that your human agents use
What WebRTC can’t do is dial a regular phone number. It has no native PSTN access. A customer on a mobile phone dialing your number is not on WebRTC. They’re on the telephone network, and reaching them requires SIP.
There’s an underappreciated upside to the screen side, though: a call that stays inside WebRTC end-to-end keeps Opus and stays wideband. That means noticeably better transcription accuracy than a phone call. We’ll unpack this in the next section.
Here’s the one-line distinction worth memorizing:
- SIP is the telephone network’s call-control layer.
- WebRTC is an internet real-time media transport.
Different layers, different jobs. So if they do different jobs, what actually happens on a single call that touches both?
How WebRTC and SIP Work Together, and the Bridge Between Them
Picture a customer dialing your business number. That call doesn’t pick SIP or WebRTC. It rides both, at different points along the path.
The mental model is simple: SIP lives at the PSTN edge, WebRTC lives inside the platform. The caller experiences neither directly, they just hear the receptionist. Pull out either layer and the system breaks. Remove SIP and the agent is unreachable by phone. Remove the internal transport and the real-time audio pipeline has no way to reach the agent.
Now picture the other front door: a visitor clicks “talk to us” on your website. There’s no phone network, no carrier, and no bridge. WebRTC carries the browser audio straight to the same AI pipeline.
Same destination, shorter path. The web door skips the carrier trunk, the media gateway, and the codec conversion entirely. The audio stays in Opus end-to-end. Both doors feed the identical STT-to-LLM-to-TTS pipeline; they just differ in how far the audio travels to get there.
This isn’t a niche architecture. Most voice AI platforms follow this pattern: SIP at the phone network edge, and WebRTC or WebSocket inside the platform. Even the model vendors have settled the debate: OpenAI’s Realtime API supports WebRTC, SIP, and WebSocket, and the guidance is to choose by where the audio originates (OpenAI).
Which means the interesting engineering isn’t choosing a protocol. It’s the bridge between them.
A third transport: WebSocket
Many voice AI deployments carry audio over WebSocket instead of WebRTC. A WebSocket is a persistent, two-way connection between two servers or between a browser and a server. Voice platforms use it to stream raw audio frames to and from agents
WebSocket has trade-offs. It runs over TCP, so a lost packet holds up every packet behind it until the retransmission arrives. WebSocket also has no built-in jitter buffer, echo cancellation, or bandwidth adaptation; your code has to handle buffering and interruptions when the caller talks over the agent. Between two servers in a data center, packet loss is rare and these costs stay small. On a customer’s mobile browser over cellular data, they show up as delays and choppy audio. That is why OpenAI recommends WebRTC for browser and mobile clients and WebSocket for server-to-server connections.
WebSocket leaves the codec ceiling unchanged. Phone audio arrives over the stream as 8 kHz G.711, the same narrowband audio described in the next section.
For a receptionist: use WebRTC for the website widget, and expect SIP plus either WebRTC or WebSocket for the phone line, depending on your platform.
How G.711 Limits Speech-to-Text Accuracy on Phone Calls
Every time a call crosses the WebRTC-to-SIP boundary, a codec conversion happens: Opus on the WebRTC side, G.711 on the PSTN side. That transcoding step isn’t free. It adds roughly 5–20 ms of latency per leg, burns CPU on every concurrent call, and is lossy in one direction: once audio is downsampled to G.711, the detail is gone for good. That matters when you’re already spending 200–500 ms per turn on speech-to-text and LLM inference.
The deeper issue is a ceiling most teams never see coming. PSTN audio is bandlimited to roughly 300 Hz–3.4 kHz, and the high-frequency cues that distinguish fricatives (the s, f, and th sounds) live above that range. On a narrowband call, that acoustic information is simply gone. Best-in-class speech-to-text tops might struggle in accuracy on 8 kHz telephony audio. A better model buys a point or two but can’t break the wall, because the limit is missing information, not model quality.
The practical consequence for a receptionist is striking: your web-widget callers get better transcription than your phone callers, purely because of the channel they arrived on. The door a caller walks through shapes how well the agent understands them.
You can’t repeal physics on the PSTN leg, but you can avoid making it worse. The validated best practice across the industry is to minimize transcoding hops and keep the highest-quality codec the whole path supports:
- Keep a single codec end-to-end where possible so no conversion happens. On a PSTN leg that’s narrowband anyway, that often means negotiating G.711 on both sides of the gateway.
- Where the entire path supports it, push your telephony provider for native Opus or wideband G.722 instead of defaulting to G.711. The wider band lifts the ASR ceiling.
- Monitor codec negotiation on the media path. Many “the agent is broken today” incidents turn out to be a carrier quietly renegotiating calls onto a lower-quality codec under congestion, showing up as degraded transcription rather than an obvious alarm.
The unifying rule: fewest conversions, highest codec the full path can carry.
Warm transfer: where both protocols meet the human
A receptionist’s whole job is routing, and the hardest moment is the handoff to a human. Do it wrong and you destroy the value the AI just created.
A blind transfer dumps the caller and forces them to repeat everything, adding 40–90 seconds of wasted handle time. A warm transfer briefs the human first, in 10–15 seconds, and preserves the context the AI already gathered. SIP REFER is the mechanism that makes the phone-side handoff work: one party redirects an active call to a third party, then steps out. Modern frameworks expose this as ready-made primitives, like LiveKit’s warm-transfer task.
Here’s the neat part: if the human agent picks up on a browser softphone, WebRTC carries their audio while SIP handles the phone-side call control. One transfer, both protocols, at the same time. This is the clearest illustration of the whole thesis: good routing isn’t a WebRTC problem or a SIP problem, it’s a both problem.
Once you see the bridge, the original question almost answers itself.
When an AI Receptionist Needs WebRTC, SIP, or Both
There is exactly one genuine fork in the road, and it has nothing to do with which protocol is “better.” It’s this: where do callers reach your agent from?
- WebRTC only. If your receptionist lives entirely on your website or app (in a widget, an in-app assistant, a browser softphone) with no phone number and no outbound dialing, you can build on WebRTC and skip SIP entirely.
- SIP required. The moment any caller dials a phone number, whether that is an inbound line, an outbound reminder, an IVR replacement, a transfer to a human’s phone, you need SIP. WebRTC alone can’t serve those callers.
- Both. Most receptionists. A business wants to be reachable on its website and by phone, so it opens both doors and uses both protocols.
| If your AI receptionist … | You need | Why |
|---|---|---|
| Only lives on your website/app | WebRTC | No phone number, no PSTN, stays wideband |
| Answers a business phone number | SIP (+ WebRTC internally) | Only SIP reaches the PSTN |
| Does outbound reminders/callbacks | SIP | WebRTC can’t dial phones; deliverability matters |
| Transfers callers to humans | SIP (+ WebRTC if the agent is on a browser) | SIP REFER carries the call control |
| Spans web + phone (typical) | Both | Each front door speaks its own protocol |
Deciding to use both is the easy part. The honest tradeoffs live in the details:
- Latency budget. Every bridge hop and codec conversion spends milliseconds against a tight turn-taking budget. Audit the full media path, not just the model.
- Channel-bound quality. Phone callers are capped by G.711’s narrowband ceiling; set transcription-accuracy expectations per channel.
- Outbound deliverability. If the receptionist dials out, STIR/SHAKEN attestation (aim for A-level) and number reputation decide whether calls get answered or land as “Spam Likely” (Softcery). This is a SIP/carrier concern with no WebRTC equivalent.
- Single-carrier risk. For a business’s main line, a one-carrier setup is a single point of failure. Multi-carrier failover keeps the receptionist from going dark during an outage.
- Build vs. buy the telephony layer. Carrier minutes are cheap (roughly $0.003–$0.014/min) next to the all-in $0.13–$0.33/min once LLM and TTS are added (Softcery). Own telephony for control and reliability, not to save pennies.
So you only “choose” if your receptionist lives in exactly one channel. Otherwise the real work isn’t the protocol — it’s designing the bridge, the carrier setup, and the handoff well. That design work is where a receptionist project actually succeeds or stalls.
Building the WebRTC-to-SIP Bridge for a Voice AI Agent
The protocol question resolves into an architecture question. WebRTC and SIP aren’t rivals to choose between. They’re adjacent layers you compose, and the engineering that matters lives in the bridge between them: media gateways, codec paths, warm-transfer orchestration, carrier selection, and failover.
At WebRTC.ventures, we live in this stack daily: WebRTC, SIP, real-time AI orchestration, media gateways, and PSTN integration. We’ve built exactly this kind of receptionist: connecting an existing PSTN number to a voice AI agent via SIP forwarding, handling the conversation, and warm-transferring to a human without porting the number or touching the carrier contract. Whether you’re adding a phone line to a web-based agent, a web widget to a phone-based agent, or building both from scratch, this is the integration layer where deployments succeed or stall.
Tell us where your callers come from, and we’ll help you design an AI receptionist that meets them there. Contact us today.
