When you deploy an AI voice agent, the protocol you choose determines whether calls connect in under 200 milliseconds or stall at two seconds. WebRTC and SIP are the two dominant transport methods for voice AI, and they solve the same problem in fundamentally different ways. WebRTC runs natively in browsers and optimises for peer-to-peer connection speed; SIP is the telecommunications industry standard that integrates with legacy phone systems and PSTN networks. Neither is universally better. The right choice depends on your call volume, existing infrastructure, acceptable latency, and whether you need PSTN fallback.
This article breaks down how each protocol works, where each excels, where each breaks down, and which scenarios fit which tool. If you are evaluating AI voice technology for your business, understanding these differences now saves you the cost of a failed deployment later.
How WebRTC and SIP Handle Voice Traffic Differently
SIP, or Session Initiation Protocol, is a signalling standard that has been the backbone of enterprise telephony since the 1990s. It is designed to establish, modify, and terminate voice calls across networks. SIP itself does not carry voice data; it tells both endpoints how to find each other and agree on a codec. Once the handshake completes, the actual audio travels over RTP, or Real-Time Transport Protocol. This separation of signalling from media is elegant for enterprise networks where a dedicated SIP trunk from a provider like Vonage or Twilio handles the routing and gatekeeping.
WebRTC, or Web Real-Time Communication, bundles signalling, media transport, and encryption into a single standard that runs in the browser or as an embedded library. When a caller initiates a WebRTC call to an AI agent, the browser itself is the phone. There is no intermediary protocol step; the browser negotiates codec, encryption, and media path directly with the other endpoint. This is faster to set up but requires both sides to speak WebRTC natively.
The practical difference: SIP requires a middleman service to route and translate. WebRTC aims to eliminate that hop. In a SIP deployment, Twilio or another carrier sits between the caller and your AI system, adding a gateway layer. In a WebRTC system, the caller's browser connects directly to your cloud infrastructure. Both work. One has lower latency potential; the other integrates with the existing telephone network.
WebRTC SIP Voice AI Latency and Call Quality
Latency is where this choice becomes operationally visible. Industry benchmarks suggest acceptable latency for conversational AI sits around 150 to 400 milliseconds round-trip. Below 150ms feels like a natural conversation; above 600ms people start to interrupt each other, and the interaction breaks down. WebRTC can achieve 150 to 300ms latency because it is designed for point-to-point calls with minimal intermediaries. The trade-off is that both endpoints need to support WebRTC natively, and if one side is a regular landline, WebRTC fails entirely.
SIP introduces latency through gateway hops. A call from a landline to your AI agent travels: landline to carrier network, carrier to SIP gateway, gateway to your AI infrastructure. Each hop adds 50 to 100ms. Operators typically report that SIP-based AI voice deployments run at 300 to 600ms latency, which is acceptable but noticeably slower. For outbound campaigns where you are dialling customers on their phones, this is usually transparent to them. For inbound calls where customers expect instant AI response, the slower responsiveness becomes noticeable after a few exchanges.
Jitter, or variation in packet arrival time, matters equally. WebRTC applies built-in adaptive bitrate control, which means it compensates for network congestion automatically and rarely produces audible artifacts. SIP delegates this to RTP and your carrier. If your SIP provider is overloaded or your internet connection is unstable, voice quality degrades visibly. WebRTC degrades more gracefully because it was designed for the public internet, not the private telephone network.
Infrastructure and Integration Requirements
SIP integration is straightforward if you already have a phone system. Most PBX systems, from Avaya to FreePBX, speak SIP natively. If you want to plug an AI voice agent into your existing contact centre, you add a SIP trunk from a carrier like Vonage, Twilio, or Bandwidth. Your phone system routes incoming calls to the AI agent, the agent picks up on the SIP trunk, processes the call, and transfers to a human or hangs up. This works because SIP is the lingua franca of enterprise telephony. Setup takes days, not weeks.
WebRTC requires you to build or buy a solution that handles browser-based calling, which means you need a signalling server to orchestrate the connection, TURN servers to relay calls if direct peer-to-peer fails, and TLS encryption everywhere. If you are running a SaaS platform where customers call your system from their web browser, WebRTC is natural; you already have the infrastructure. If you are a contact centre receiving calls from the public telephone network, WebRTC forces you to either accept only WebRTC callers or run a SIP-to-WebRTC gateway, which defeats its speed advantage.
Sysevo integrates with both protocols; the platform abstracts this choice away for most users, but understanding the underlying transport matters when you debug call quality issues or plan for scale. If your callers are predominantly on regular phones, SIP is the practical choice. If your callers are on web or mobile apps, WebRTC wins. If you need both, you run dual infrastructure, which adds cost and operational complexity.
Cost Structure and Scaling Economics
SIP costs are transparent and predictable. A dedicated SIP trunk from Twilio runs 35 to 50 cents per minute for inbound calls, plus a monthly trunk fee of 25 to 50 dollars. If you run 10,000 inbound minutes per month (about 330 minutes per day), that costs 350 to 500 dollars monthly plus trunk fees. This scales linearly with volume. Carriers compete on price, so the market is commoditised. A mid-market contact centre handling 100,000 minutes monthly spends 3,500 to 5,000 dollars on SIP alone, which is a known budget line.
WebRTC costs are inverted. You pay for infrastructure, not per-minute usage. Running a WebRTC signalling server with TURN relays on AWS costs 100 to 300 dollars per month for moderate volume, then scales with compute. At 10,000 minutes of calls, your infrastructure cost is negligible; at 1 million minutes monthly, you might spend 500 to 1,500 dollars. The crossover point is around 2 million minutes per month. Below that, WebRTC is cheaper. Above that, SIP becomes cheaper. For most small-to-medium businesses, SIP is the economically rational choice because the call volume never justifies WebRTC infrastructure investment.
There is a hidden cost to WebRTC: you own the uptime. A SIP carrier guarantees 99.9% availability; if they go down, you call them, not the other way around. With WebRTC, your TURN servers are your responsibility. You need redundancy, failover, and monitoring. This is a 20-30% operational overhead that SIP pushes onto the carrier. For businesses with small ops teams, this is a material cost.
Integration with Existing Phone Systems and PSTN
PSTN, the Public Switched Telephone Network, is still the way 80 to 90% of business calls arrive. Your customers call a phone number, which terminates on the PSTN, which routes to your AI agent. This requires SIP or another PSTN-compatible protocol. WebRTC cannot reach the PSTN directly. If you want to use WebRTC, you must accept calls only from WebRTC callers (customers on your website or app) or run a SIP-to-WebRTC gateway. This gateway adds latency, cost, and failure points, negating most of the speed benefit.
SIP is the glue between PSTN and your AI infrastructure. Every VOIP provider bridges this gap using SIP. Your phone number lives on the PSTN, signals arrive as SIP, your AI agent processes them, and responses travel back the same route. This is battle-tested infrastructure. Thousands of contact centres depend on it daily. The protocol is mature enough that failure is rare; when it happens, the problem is usually your internet connection or your provider's gateway, not the protocol itself.
If you need two-way integration, where your AI agent makes outbound calls to PSTN numbers (customers, partners, follow-ups), SIP is non-negotiable. WebRTC simply cannot initiate calls to a landline. Outbound campaigns, which are a core use case for AI voice, require SIP. This is one of the clearest decision factors. If outbound calling is in your roadmap, rule out WebRTC unless you plan to call only WebRTC-enabled endpoints.
Where Each Protocol Struggles
WebRTC excels at low-latency, encrypted, peer-to-peer calls where both endpoints are modern and connected. It fails immediately if one endpoint is a legacy phone system, a PSTN number, or a network that does not support UDP traffic. Many corporate firewalls and mobile networks block or throttle UDP, which is what WebRTC uses for media. If your customers are calling from offices with strict firewall policies, WebRTC calls will fail or fall back to TCP, which adds latency. SIP has these problems too, but carriers spend decades optimizing firewalls for SIP traffic specifically.
SIP, by contrast, works everywhere the telephone network works. It integrates seamlessly with legacy PBX systems, PSTN gateways, and carrier networks. But it introduces latency through gateways, costs you per-minute fees that scale with volume, and requires a carrier relationship. If you want low-latency, direct calls where you control all the infrastructure, SIP feels inefficient. You pay for every minute even if your network infrastructure is idle. For a contact centre running campaigns where every minute costs, this adds up.
Neither protocol is wrong. SIP is wrong if you only accept calls from WebRTC browsers and want to minimise carrier fees. WebRTC is wrong if your callers are on regular phones. The mistake is choosing based on speed alone; the right choice accounts for who your callers are, what your infrastructure already looks like, and how many calls you expect to handle.
Choosing the Right Protocol for Your AI Voice Stack
Start with the question: where do your callers come from? If they are calling from regular phones, landlines, or mobile networks, SIP is mandatory. WebRTC is a nice-to-have for browser or app-based calling, not a replacement. If your entire call base comes from a web widget on your website, WebRTC is the natural choice. Most businesses fall in the middle: some inbound from the PSTN, some from web, some from mobile app. In that case, you support both. SIP handles the PSTN; WebRTC handles the web and app callers. Your AI agent infrastructure accepts both simultaneously.
Second question: what is your call volume now, and what do you expect in 12 months? Under 50,000 monthly minutes, SIP is cheaper and simpler. Over 500,000 monthly minutes, WebRTC infrastructure becomes economically competitive with SIP per-minute fees. The break-even point varies with your carrier; Twilio is more expensive than Bandwidth, which changes the math. If you cannot answer this question with data, default to SIP. It is the lower-risk choice for unknown volume.
Third: do you have existing infrastructure that expects SIP? If you have a PBX, a contact centre, or a VOIP system already running SIP trunks, adding an AI voice agent means connecting to your existing SIP infrastructure. This is a day or two of work. Switching to WebRTC means ripping out your existing architecture and building something new. Unless your current setup is actively failing you, the switching cost is not justified. When evaluating voice AI solutions, ask whether they support your existing SIP infrastructure natively.
Implementation Considerations and Real-World Trade-offs
A real scenario: a mid-market B2B SaaS company with a 10-person sales team wants to deploy AI voice to qualify inbound leads. Their callers are warm prospects coming from email campaigns, landing pages, and partners. About 60% call from regular phones; 40% call from a web widget on the website. Their current phone system is a cloud-based PBX with SIP trunking through Twilio, costing them 800 dollars per month. They expect 5,000 inbound calls per month initially, scaling to 20,000 within 18 months.
For this company, SIP is the rational choice. The Twilio SIP trunk they already have can route calls to the AI agent without additional configuration. The agent processes the call, captures the caller's intent, writes it to a CRM system, and transfers warm prospects to a human. Cost: roughly 250 to 300 dollars per month for the AI agent infrastructure, plus their existing Twilio SIP spend. If they tried to build WebRTC infrastructure, they would need TURN servers, signalling orchestration, and monitoring. The development cost would be 10,000 to 15,000 dollars, and the ongoing operational cost would be 200 to 400 dollars per month. They would save 50 to 100 dollars monthly but spend months building and maintaining it. SIP wins.
Contrast this with a B2C customer service platform that lives entirely on web. Customers call from the website widget. There is no PSTN involvement. Here, WebRTC is more efficient. The company builds a signalling server, runs TURN relays, and routes calls directly to the AI agent. Latency is 200 to 300ms instead of 400 to 600ms. Customers experience faster AI response. The cost is the infrastructure, which scales more efficiently than per-minute SIP fees at high volume. WebRTC wins. The two companies need different protocols because they have different callers and different economics.
Frequently Asked Questions
Can I use both WebRTC and SIP with a single AI voice agent?
Yes. Your AI infrastructure accepts calls via whichever protocol the incoming call uses. A SIP call arrives from your carrier; a WebRTC call arrives from your website. The agent processes both identically. In practice, you run one signalling layer that speaks both protocols, or you run two systems that feed into one agent. Most custom solutions support this hybrid approach, though it adds complexity.
Does WebRTC work with phone numbers and the PSTN?
WebRTC cannot initiate or receive calls on PSTN phone numbers directly. If you need a phone number your customers dial, you need SIP or another PSTN-compatible protocol. WebRTC works for web-to-web or app-to-app calling only. To bridge the gap, you run a SIP gateway that converts PSTN calls to WebRTC, but this adds latency and defeats the speed advantage.
Which protocol is more secure?
WebRTC includes mandatory encryption (SRTP and DTLS). SIP does not encrypt by default, though SIPS (SIP over TLS) and encrypted RTP add security. For security-sensitive applications, both can be configured securely. WebRTC has a slight edge because encryption is built-in; SIP requires explicit configuration.
Will a SIP trunk work with my existing PBX and an AI voice agent?
Yes, in most cases. Your existing SIP trunk can route calls to the AI agent instead of or before routing to a human queue. This is the simplest integration path if you already have SIP infrastructure. Verify with your AI vendor that they support SIP trunks from your carrier.
At what call volume does WebRTC become cheaper than SIP?
Typically around 1 to 2 million minutes per month, depending on your SIP carrier rates. Below that, SIP per-minute fees are cheaper than WebRTC infrastructure costs. At very high volumes, WebRTC can save 30 to 50% on telephony costs, but you must own the infrastructure and operations overhead.
What happens if my WebRTC infrastructure fails?
Your calls stop. With SIP, your carrier handles failover and redundancy as part of their SLA. With WebRTC, you own uptime. You need redundant TURN servers, monitoring, and a failover plan. This is not a reason to avoid WebRTC, but it is a cost that SIP hides from you.
If you are ready to evaluate an AI voice solution and want to understand how these protocols fit your specific infrastructure, book a call with a specialist. They can assess your current setup, estimate call volume, and recommend the protocol that minimises your costs and maximises call quality for your use case.
