- Both intend to support the creation of media sessions between two IP connected endpoints and both use SDP
- SIP strength is in connecting to the telecom world, WebRTC strength is in the internet world
- SIP is a signaling protocol. WebRTC needs one but doesn’t have one defined . It can use SIP or other signaling protocols
- WebRTC mandates the used of some SIP optional features: SRTP, DTLS, specific codecs, etc.
Interoperability differences between SIP and WebRTC
| Item | SIP | WebRTC |
|---|---|---|
| Media transport | RTP, SRTP (opt) | SRTP, new RTP Profiles |
| Session Negotiation | SDP, offer/answer | SDP trickle |
| NAT traversal | STUN TURN ICE | ICE (include STUN/TURN) |
| Media transport | Separate : audio/video, RTP vs RTCP | Same path with all media and control |
| Security Model | User trusts device & service provider | User trusts browser but not website |
| Audio Codecs | Typically G.711. G.729 G.722 Speex | Mandatory Opus & G.711, optionally others |
| Video Codecs | Typically H261 H263 H264 | Undefined yet but likely VP8 and or H264 |