Got it, since you already have a working WebRTC-to-Android WebRTC call system, extending it to support calls to phone numbers or PSTN is totally feasible. Below’s a practical, step-by-step breakdown to help you build this out:

Core Concept First #

WebRTC doesn’t natively support PSTN connectivity—you’ll need an intermediate layer (either a self-hosted gateway or a third-party service) to bridge WebRTC’s media streams with the traditional telephone network. This layer handles converting WebRTC’s RTP packets into PSTN-compatible signals and vice versa.

Option 1: Use a Third-Party PSTN Service (Fastest Integration) #

If you want to avoid the overhead of maintaining your own infrastructure, this is the way to go. Most providers offer pre-built SDKs and APIs that play nicely with existing WebRTC setups.

• Pick a provider: Look for services with WebRTC-to-PSTN support. Sign up, get your API credentials, and rent a virtual phone number if needed.
• Integrate their SDK into your clients:
  • For your web client: Add the provider’s WebRTC SDK alongside your existing WebRTC code. When a user enters a phone number, trigger the provider’s API to initiate a call—they’ll handle the WebRTC-to-PSTN bridging.
  • For your Android client: Use the provider’s Android SDK (or wrap their Web SDK in a WebView if preferred) to add PSTN call functionality, aligning it with your existing WebRTC call UI/UX.
• Sync call states: Listen for events like call initiation, connection, hangup, and failures from the provider’s SDK, and sync these with your app’s existing call state management (so users see consistent feedback whether calling another WebRTC client or a phone number).
• Test thoroughly: Use the provider’s test numbers first to validate call quality, then move to real numbers across different carriers.

Option 2: Self-Host a SIP Gateway (Full Control) #

If you need full control over your infrastructure or have specific compliance requirements, setting up a SIP gateway is the right choice.

• Choose and deploy a gateway: Popular open-source options are Asterisk or FreeSWITCH. Install it on a server (cloud or on-prem) and configure a SIP trunk—this is your link to the PSTN (you’ll need to partner with a SIP trunk provider or a telecom carrier to get this access).
• Bridge WebRTC and SIP:
  • Configure the gateway to accept WebRTC connections (most gateways support this out of the box with minimal setup).
  • Set up routing rules: When your WebRTC client sends a call request to a phone number, the gateway translates it into a SIP call and routes it through the SIP trunk to the PSTN.
• Update your clients:
  • For web/Android clients, add SIP call logic using libraries like PjSIP (for Android) or SIP.js (for web). This lets your clients communicate directly with the SIP gateway to initiate PSTN calls.
  • Ensure your client can handle both WebRTC-to-WebRTC and WebRTC-to-SIP/PSTN calls with a unified interface.
• Optimize media handling: Make sure the gateway converts between WebRTC’s preferred codecs (like Opus) and PSTN-compatible codecs (like G.711). Adjust settings for latency and jitter to maintain call quality.
• Monitor and maintain: Set up logging, monitoring, and backup systems to keep the gateway running reliably. You’ll also need to handle billing and compliance on your end.

Key Considerations to Keep in Mind #

• Compliance: Different regions have strict rules for PSTN calls (e.g., caller ID verification, data privacy laws like GDPR). Make sure your setup adheres to local regulations.
• Call quality: PSTN calls have different audio characteristics than WebRTC—test across networks and carriers to address issues like echo, delay, or dropped calls.
• Billing transparency: If using a third-party service, understand their pricing model (per-minute rates, setup fees). For self-hosted, track usage to manage costs with your SIP trunk provider.
• Error handling: Add clear user feedback for common issues like invalid phone numbers, network failures, or gateway downtime.

内容的提问来源于stack exchange，提问作者aadhi