diff --git a/404.html b/404.html index 66c6dcb91..0dd90769e 100644 --- a/404.html +++ b/404.html @@ -10,7 +10,7 @@ - +
diff --git a/assets/js/144994e6.12bed236.js b/assets/js/144994e6.fd76df99.js similarity index 99% rename from assets/js/144994e6.12bed236.js rename to assets/js/144994e6.fd76df99.js index 36e0e5e59..c49b61efd 100644 --- a/assets/js/144994e6.12bed236.js +++ b/assets/js/144994e6.fd76df99.js @@ -1 +1 @@ -"use strict";(self.webpackChunkroutr_docs=self.webpackChunkroutr_docs||[]).push([[1171],{8162:(e,n,t)=>{t.r(n),t.d(n,{assets:()=>c,contentTitle:()=>i,default:()=>d,frontMatter:()=>r,metadata:()=>a,toc:()=>l});var s=t(4848),o=t(8453);const r={},i="RTPRelay",a={id:"development/components/rtprelay",title:"RTPRelay",description:"The RTPRelay is an optional middleware service that can control RTPEngine instances. The RTPRelay component enables interoperability between WebRTC-based clients, such as SIP.js, and legacy SIP clients. Another use case for the RTPRelay is to help SIP clients who cannot send and receive media directly.",source:"@site/docs/development/components/rtprelay.md",sourceDirName:"development/components",slug:"/development/components/rtprelay",permalink:"/docs/2.0.0/development/components/rtprelay",draft:!1,unlisted:!1,editUrl:"https://github.com/fonoster/routr-website/edit/main/docs/development/components/rtprelay.md",tags:[],version:"current",frontMatter:{},sidebar:"tutorialSidebar",previous:{title:"Requester",permalink:"/docs/2.0.0/development/components/requester"},next:{title:"APIServer",permalink:"/docs/2.0.0/development/components/apiserver"}},c={},l=[{value:"Configuration Spec",id:"configuration-spec",level:2},{value:"Communication and Protobuf Spec",id:"communication-and-protobuf-spec",level:2},{value:"Launching the RTPRelay with Docker",id:"launching-the-rtprelay-with-docker",level:2},{value:"Quick Test with gRPCurl",id:"quick-test-with-grpcurl",level:2}];function h(e){const n={a:"a",code:"code",h1:"h1",h2:"h2",li:"li",p:"p",pre:"pre",ul:"ul",...(0,o.R)(),...e.components};return(0,s.jsxs)(s.Fragment,{children:[(0,s.jsx)(n.h1,{id:"rtprelay",children:"RTPRelay"}),"\n",(0,s.jsx)(n.p,{children:"The RTPRelay is an optional middleware service that can control RTPEngine instances. The RTPRelay component enables interoperability between WebRTC-based clients, such as SIP.js, and legacy SIP clients. Another use case for the RTPRelay is to help SIP clients who cannot send and receive media directly."}),"\n",(0,s.jsx)(n.h2,{id:"configuration-spec",children:"Configuration Spec"}),"\n",(0,s.jsx)(n.p,{children:"Unlike other components, the RTPRelay service does not have a configuration file. However, the following environment is to configure the service:"}),"\n",(0,s.jsxs)(n.ul,{children:["\n",(0,s.jsxs)(n.li,{children:[(0,s.jsx)(n.code,{children:"BIND_ADDR"})," - The IP address and port to bind the gRPC server. Defaults to ",(0,s.jsx)(n.code,{children:"0.0.0.0:51903"})]}),"\n",(0,s.jsxs)(n.li,{children:[(0,s.jsx)(n.code,{children:"RTPENGINE_HOST"})," - The IP address or hostname of the RTPEngine service. Required."]}),"\n",(0,s.jsxs)(n.li,{children:[(0,s.jsx)(n.code,{children:"RTPENGINE_PORT"})," - The port of the RTPEngine service. Defaults to ",(0,s.jsx)(n.code,{children:"22222"}),"."]}),"\n",(0,s.jsxs)(n.li,{children:[(0,s.jsx)(n.code,{children:"RTPENGINE_TIMEOUT"})," - The timeout in milliseconds for the RTPEngine service. Defaults to ",(0,s.jsx)(n.code,{children:"5000"}),"."]}),"\n"]}),"\n",(0,s.jsx)(n.h2,{id:"communication-and-protobuf-spec",children:"Communication and Protobuf Spec"}),"\n",(0,s.jsx)(n.p,{children:'Services communicate with RTPRelay service using gRPC. The RTPRelay, in turn, communicates with RTPEngine using the "ng protocol." The contract for communication with RTPRelay service consists of the following protobuf:'}),"\n",(0,s.jsx)(n.pre,{children:(0,s.jsx)(n.code,{className:"language-protobuf",children:'syntax = "proto3";\n\npackage fonoster.routr.processor.v2beta1;\n\nimport "common.proto";\nimport "sipmessage.proto";\n\n// Processor service\nservice Processor {\n // Process Message Request\n rpc ProcessMessage (MessageRequest) returns (MessageResponse) {}\n}\n\nenum Method {\n UNKNOWN = 0;\n // Communicates user location (hostname, IP)\n REGISTER = 1;\n // Establishes a session\n INVITE = 2;\n // Transports Instant Messages\n MESSAGE = 3;\n // Publishes an event to the Server\n PUBLISH = 4;\n // Notifies the subscriber of a new event\n NOTIFY = 5;\n // Subscribes for Notification from the notifier\n SUBSCRIBE = 6;\n // Confirms an INVITE request\n ACK = 7;\n // Terminates a session\n BYE = 8;\n // Cancels establishing of a session\n CANCEL = 9;\n // Communicates information about the capabilities of calling and receiving SIP phones\n OPTIONS = 10;\n // Provisional Acknowledgement\n PRACK = 11;\n // Sends mid-session information\n INFO = 12;\n // Asks the recipient to issue a call transfer\n REFER = 13;\n // Modifies the state of a session\n UPDATE = 14;\n}\n\nmessage NetInterface {\n string host = 1;\n int32 port = 2;\n fonoster.routr.common.v2beta1.Transport transport = 3;\n}\n\nmessage MessageRequest {\n // Same as the Call-Id header \n string ref = 1;\n string edge_port_ref = 2;\n Method method = 3;\n NetInterface sender = 4;\n repeated NetInterface listening_points = 5;\n repeated string external_addrs = 6;\n repeated string localnets = 7;\n fonoster.routr.sipmessage.v2beta1.SIPMessage message = 8;\n mapOne easy way to interact with the RTPRelay for testing and development is to use gRPCurl. The following example shows how to send a SIP Message to the RTPRelay.
-grpcurl -plaintext \
-import-path /path/to/protos \
-proto processor.proto -d '{...}' \
localhost:51901 \
fonoster.routr.processor.v2beta1.Processor/ProcessMessage
grpcurl -plaintext \
-import-path /path/to/protos \
-proto processor.proto -d '{...}' \
localhost:51903 \
fonoster.routr.processor.v2beta1.Processor/ProcessMessage