A lightweight sip proxy, location server, and registrar for a reliable and scalable SIP infrastructure.
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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.
We are building Routr in the open. The best to communicate with us via GitHub Discussions.
Special Announcement:
We now have a Discord Channel
There we plan to discuss roadmaps, feature requests, and more
Join us today
Twitter: @fonoster
Issue tracker: Use the GitHub issue tracker for the various Routr repositories to file bugs and features request. If you need support, please send your questions to the routr-users mailing list 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.
Routr's main features are:
To learn more, read the documentation.
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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: v2draft1
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: v2draft1
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.
For a quick demo of Routr, follow the next two steps:
➊ Clone the repository and run the server
git clone https://github.com/fonoster/routr
docker-compose up
➋ 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).
Routr's one-click interactive deployment will familiarize you with the server in development mode.
Deploying Routr in Kubernetes is coming soon.
For bugs, questions, and discussions, please use the Github Issues
For contributing, please see the following links:
Sponsors
We're glad to be supported by respected companies and individuals from several industries.
Find all our supporters here
Copyright (C) 2022 by Fonoster Inc. MIT License (see LICENSE for details).
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