Routr (v2)

Routr is a lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators.

Community

We are building Routr in the open. The best to communicate with us via GitHub Discussions.

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^{Special Announcement:}

_{We now have a Discord Channel
There we plan to discuss roadmaps, feature requests, and more
Join us today}

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Twitter: @fonoster

Issue tracker: Use the GitHub issue tracker to file bugs and features request. If you need support, please start a Discussion rather than filing a GitHub issue.

Please do not ask individual project members for support. Use the channels above instead, where the whole community can help you and benefit from the solutions provided. Please contact us for Commercial Support if you need more than community support.

Features

Routr's main features are:

• Common SIP Server functions; Proxy, Registrar, Location Service
• Programmable Routing
• Load balancing strategies against Media Servers like Asterisk and FreeSWITCH
• Session Affinity
• Multi-Tenant/Multi-Domain with Domain level Access Control List
• Region-based routing
• Configurable routing strategies; Intra-Domain, Domain Ingress, Domain Egress, and Peer Egress
• No single point of failure
• Transport: TCP, UDP, TLS, WS, WSS
• In-memory and Redis Location Service
• JSON and YAML files as a data source
• Postgres as a data source
• Server management with the gRPC API
• NodeSDK
• Command-Line Tool
• RTPEngine Middleware
• Helm Chart for Kubernetes Deployments
• Endpoint Authentication with JWT (For web phones)
• Support for STIR/SHAKEN
• Web Application

To learn more, read the documentation. 📚

Give a Star! ⭐

If you like this project or plan to use it in the future, please give it a star. Thanks 🙏

Example configuration

Consider a situation where you want to deploy the server and send all PSTN traffic to a conference room in Asterisk. For such a scenario, you must configure a Peer to present your feature server and a Number to route calls from the PSTN.

First, start by creating a Peer configuration for your Asterisk server similar to the following one:

apiVersion: v2beta1
kind: Peer
ref: peer-01
metadata:
  name: Asterisk (Media Server)
spec:
  aor: backend:conference
  username: asterisk
  credentialsRef: credentials-01
  loadBalancing:
    withSessionAffinity: true
    algorithm: least-sessions

Notice that the loadBalancing section sets the withSessionAffinity to true. We need session affinity to ensure that all calls related to the conference arrive on the same Asterisk server. Every Asterisk server that registers using the asterisk username will be grouped under the backend:conference Address of Record (AOR).

Next, we need to tell Routr to map all inbound calls from a given Number to the conference room in Asterisk. For that, we use the aorLink and sessionAffinityHeader on the desired Number. Here is an example:

apiVersion: v2beta1
kind: Number
ref: number-01
metadata:
  name: "(706)604-1487"
  geoInfo:
    city: Columbus, GA
    country: USA
    countryISOCode: US
spec:
  trunkRef: trunk-01
  location:
    telUrl: tel:+17066041487
    aorLink: backend:conference
    sessionAffinityHeader: X-Room-Id
    extraHeaders:
      # Appends the X-Room-Id header to all inbound calls
      - name: X-Room-Id
        value: jsa-shqm-iyo

The last scenario is one of the many possible scenarios you can accomplish with Routr (v2). Please spend some time getting familiar with the configuration files.

Deployment

Instant Server deployment with Docker and Compose

For a quick demo of Routr, follow the next two steps:

1. Clone the repository and run the server

git clone https://github.com/fonoster/routr
docker-compose up 

2. Connect to Routr using Zoiper or another softphone

In the config/resources, you will find the domains.yaml and agents.yaml files. Those files contain the configuration to run a simple local network with two SIP Agents (John and Jane).

Deploying in development mode with Gitpod

Routr's one-click interactive deployment will familiarize you with the server in development mode.

To connect to your instance, follow these steps:

First, add your public SSH-keys to your Gitpod account by going to Gitpod account keys and adding your public key.

Next, find your Gitpod workspace and click on the "More" button. Then, select "Connect via SSH."

Finally, copy the SSH Command and run it in your terminal by pasting it and pressing Enter. The command should look like this:

ssh -L 5060:localhost:5060 <workspace-ssh-connection>

Replace with your own workspace SSH connection.

For example, your command might look like this:

ssh -L 5060:localhost:5060 fonoster-routr-mn8nsx0d9px@fonoster-routr-mn8nsx0d9px.ssh.ws-us90.gitpod.io

This command forwards traffic from your local port 5060 to your Gitpod workspace's port 5060, allowing you to access your instance.

Kubernetes

Deploying Routr in Kubernetes is coming soon.

Bugs and Feedback

For bugs, questions, and discussions, please use the Github Issues

Contributing

For contributing, please see the following links:

• Contribution Documents
• Contributors

Sponsors

We're glad to be supported by respected companies and individuals from several industries.

Find all our supporters here

Become a Github Sponsor

Authors

• Pedro Sanders

License

Copyright (C) 2023 by Fonoster Inc. MIT License (see LICENSE for details).