[{"data":1,"prerenderedAt":400},["ShallowReactive",2],{"post-\u002Fblog\u002Fwebrtc-vs-sip-trunking":3},{"id":4,"title":5,"author":6,"body":7,"category":34,"coverImage":384,"date":385,"description":386,"extension":387,"meta":388,"navigation":389,"path":390,"readingTime":391,"seo":392,"stem":393,"tags":394,"__hash__":399},"posts\u002Fblog\u002Fwebrtc-vs-sip-trunking.md","When to choose WebRTC vs SIP trunking for your voice app","Tumarm Engineering",{"type":8,"value":9,"toc":370},"minimark",[10,14,18,25,30,36,52,59,65,76,81,85,88,93,101,106,117,122,130,135,143,147,152,163,166,170,176,179,183,189,192,196,202,205,209,340,344,347,350,353,356,359],[11,12,5],"h1",{"id":13},"when-to-choose-webrtc-vs-sip-trunking-for-your-voice-app",[15,16,17],"p",{},"Teams building voice into their applications usually frame this as \"WebRTC vs SIP\" — but that's not quite the right question. WebRTC is a browser\u002Fapp protocol for real-time media. SIP trunking is a carrier-connectivity model. You can have both, and for most serious voice applications, you will.",[15,19,20,21],{},"The real question is: ",[22,23,24],"strong",{},"which model carries which leg of the call, and what does that mean for your architecture?",[26,27,29],"h2",{"id":28},"what-each-technology-actually-does","What each technology actually does",[15,31,32,35],{},[22,33,34],{},"WebRTC"," is an open standard for real-time audio, video, and data in browsers and native apps. It handles:",[37,38,39,43,46,49],"ul",{},[40,41,42],"li",{},"Peer-to-peer and SFU-based media between endpoints",[40,44,45],{},"NAT traversal via ICE\u002FSTUN\u002FTURN",[40,47,48],{},"Media security via DTLS-SRTP (mandatory)",[40,50,51],{},"Codec negotiation (Opus, VP8\u002FVP9\u002FAV1, G.711)",[15,53,54,55,58],{},"It does ",[22,56,57],{},"not"," handle PSTN connectivity. A WebRTC call between two browser tabs works. A WebRTC call from a browser tab to a landline requires a WebRTC-to-SIP gateway.",[15,60,61,64],{},[22,62,63],{},"SIP trunking"," is a way to connect your phone system to the public telephone network (PSTN). It handles:",[37,66,67,70,73],{},[40,68,69],{},"PSTN termination and origination",[40,71,72],{},"Number (DID) management",[40,74,75],{},"Compliance requirements (STIR\u002FSHAKEN, CNAM, E911)",[15,77,54,78,80],{},[22,79,57],{}," define the client protocol. Your clients can be SIP hardphones, WebRTC browsers, or proprietary apps — SIP trunking is about the carrier leg, not the endpoint leg.",[26,82,84],{"id":83},"the-decision-framework","The decision framework",[15,86,87],{},"Answer these four questions:",[15,89,90],{},[22,91,92],{},"1. Do your users need to call or receive calls from regular phone numbers?",[37,94,95,98],{},[40,96,97],{},"Yes → you need SIP trunking, regardless of what your app endpoints use",[40,99,100],{},"No (app-to-app only) → SIP trunking is optional",[15,102,103],{},[22,104,105],{},"2. Where are your users?",[37,107,108,111,114],{},[40,109,110],{},"In a browser or mobile app → WebRTC is the right endpoint protocol",[40,112,113],{},"On a desk phone or SIP softphone → native SIP works; WebRTC adds complexity for no gain",[40,115,116],{},"Mixed → you'll need both, bridged at a media gateway",[15,118,119],{},[22,120,121],{},"3. What are your latency and quality requirements?",[37,123,124,127],{},[40,125,126],{},"Sub-200ms round-trip, high quality → WebRTC with Opus, direct P2P or SFU",[40,128,129],{},"\"Phone quality\" acceptable → G.711 SIP is fine; Opus via WebRTC is better but overkill",[15,131,132],{},[22,133,134],{},"4. Do you have regulatory requirements (E911, recording, STIR\u002FSHAKEN)?",[37,136,137,140],{},[40,138,139],{},"Yes → SIP trunking with a certified SBC; WebRTC-only architectures don't expose these hooks easily",[40,141,142],{},"No → either works",[26,144,146],{"id":145},"common-architectures","Common architectures",[148,149,151],"h3",{"id":150},"app-to-app-only-no-pstn","App-to-app only (no PSTN)",[153,154,159],"pre",{"className":155,"code":157,"language":158},[156],"language-text","Browser\u002FApp → WebRTC → SFU (LiveKit) → Browser\u002FApp\n","text",[160,161,157],"code",{"__ignoreMap":162},"",[15,164,165],{},"Use when: team chat, video conferencing, in-app voice, gaming comms. PSTN never enters the picture. WebRTC handles everything. Cost: SFU infrastructure + TURN servers.",[148,167,169],{"id":168},"app-to-pstn-hybrid","App-to-PSTN (hybrid)",[153,171,174],{"className":172,"code":173,"language":158},[156],"Browser\u002FApp → WebRTC → Gateway (FreeSWITCH\u002FKamailio) → SIP Trunk → PSTN\n",[160,175,173],{"__ignoreMap":162},[15,177,178],{},"Use when: click-to-call from a web app, browser-based contact center agents, customer support tools. The gateway handles the WebRTC-to-SIP protocol translation and the SRTP-to-RTP encryption translation. This is the most common architecture for customer-facing voice apps.",[148,180,182],{"id":181},"pstn-to-pstn-with-webrtc-monitoring","PSTN-to-PSTN with WebRTC monitoring",[153,184,187],{"className":185,"code":186,"language":158},[156],"Phone → SIP Trunk → SBC → FreeSWITCH → SIP Trunk → Phone\n                              ↓\n                     WebRTC monitoring tap\n",[160,188,186],{"__ignoreMap":162},[15,190,191],{},"Use when: contact centers that want browser-based supervisor tools listening to SIP calls. The core call path is SIP; WebRTC is a tap layer, not the primary transport.",[148,193,195],{"id":194},"pure-sip-no-webrtc","Pure SIP (no WebRTC)",[153,197,200],{"className":198,"code":199,"language":158},[156],"SIP Phone → SBC → PBX → SIP Trunk → PSTN\n",[160,201,199],{"__ignoreMap":162},[15,203,204],{},"Use when: enterprise PBX replacement, desk phone deployments, traditional carrier services. If your users are on desk phones and your use case is traditional telephony, adding WebRTC introduces complexity with no user-facing benefit.",[26,206,208],{"id":207},"comparison-table","Comparison table",[210,211,212,227],"table",{},[213,214,215],"thead",{},[216,217,218,222,224],"tr",{},[219,220,221],"th",{},"Dimension",[219,223,34],{},[219,225,226],{},"SIP Trunking",[228,229,230,242,253,264,274,285,296,307,318,329],"tbody",{},[216,231,232,236,239],{},[233,234,235],"td",{},"Browser\u002Fapp native",[233,237,238],{},"✅ Yes",[233,240,241],{},"❌ Requires SIP stack",[216,243,244,247,250],{},[233,245,246],{},"PSTN connectivity",[233,248,249],{},"❌ Requires gateway",[233,251,252],{},"✅ Native",[216,254,255,258,261],{},[233,256,257],{},"E911 compliance",[233,259,260],{},"❌ Complex",[233,262,263],{},"✅ Standard",[216,265,266,269,272],{},[233,267,268],{},"STIR\u002FSHAKEN",[233,270,271],{},"❌ N\u002FA",[233,273,263],{},[216,275,276,279,282],{},[233,277,278],{},"Call recording (legal)",[233,280,281],{},"⚠️ App-layer",[233,283,284],{},"✅ Network-layer",[216,286,287,290,293],{},[233,288,289],{},"Codec flexibility",[233,291,292],{},"✅ Opus, VP8, etc.",[233,294,295],{},"⚠️ G.711\u002FG.729 typical",[216,297,298,301,304],{},[233,299,300],{},"NAT traversal",[233,302,303],{},"✅ ICE\u002FSTUN\u002FTURN",[233,305,306],{},"⚠️ Requires SBC",[216,308,309,312,315],{},[233,310,311],{},"Encryption",[233,313,314],{},"✅ DTLS-SRTP mandatory",[233,316,317],{},"⚠️ SRTP optional",[216,319,320,323,326],{},[233,321,322],{},"Desk phone support",[233,324,325],{},"❌",[233,327,328],{},"✅",[216,330,331,334,337],{},[233,332,333],{},"Setup complexity",[233,335,336],{},"Medium",[233,338,339],{},"Low–Medium",[26,341,343],{"id":342},"what-most-teams-actually-ship","What most teams actually ship",[15,345,346],{},"A WebRTC front-end (browser or mobile app) bridged to a SIP trunk via a media gateway. The gateway is FreeSWITCH or Asterisk for simpler cases, Kamailio + rtpengine for high-volume carrier-grade work.",[15,348,349],{},"The WebRTC side gives you: a great browser experience, Opus codec quality, mandatory encryption, and no SIP client to install.",[15,351,352],{},"The SIP trunk side gives you: PSTN access, real phone numbers, E911, STIR\u002FSHAKEN attestation, and carrier-grade reliability for the PSTN leg.",[15,354,355],{},"The gateway in between handles: SRTP↔RTP transcoding, SDP negotiation differences, codec normalization, and the signaling protocol translation (SIP-over-WebSocket on the WebRTC side, UDP\u002FTCP SIP on the trunk side).",[15,357,358],{},"If you're building a new voice application today:",[37,360,361,364,367],{},[40,362,363],{},"If you have browser or mobile app endpoints → use WebRTC on the client side",[40,365,366],{},"If you need PSTN access → add SIP trunking on the carrier side",[40,368,369],{},"If you need both → deploy a media gateway. This is the right call 90% of the time, not a compromise.",{"title":162,"searchDepth":371,"depth":371,"links":372},2,[373,374,375,382,383],{"id":28,"depth":371,"text":29},{"id":83,"depth":371,"text":84},{"id":145,"depth":371,"text":146,"children":376},[377,379,380,381],{"id":150,"depth":378,"text":151},3,{"id":168,"depth":378,"text":169},{"id":181,"depth":378,"text":182},{"id":194,"depth":378,"text":195},{"id":207,"depth":371,"text":208},{"id":342,"depth":371,"text":343},"https:\u002F\u002Fimages.unsplash.com\u002Fphoto-1451187580459-43490279c0fa?w=1200&q=80","2025-09-10","A decision framework for product teams choosing between WebRTC and SIP trunking — including architecture tradeoffs, use cases, and a comparison table.","md",{},true,"\u002Fblog\u002Fwebrtc-vs-sip-trunking",7,{"title":5,"description":386},"blog\u002Fwebrtc-vs-sip-trunking",[395,396,397,398],"webrtc","sip-trunking","architecture","voip","BgVCYSOpIlMl2ZrmOYemWo_eFIdT3hIeqxQgEndqHTg",1776964996445]