Software Engineer - WebRTC & VoIP

This listing is synced directly from the company ATS.

Role Overview

As a WebRTC Software Engineer at SignalWire, you will be responsible for adding new features to critical communication products and providing Tier 3 support for VoIP and WebRTC issues. This is a mid-level position that requires hands-on experience with C programming and specific VoIP/WebRTC applications. Your contributions will directly impact the success of SignalWire's European expansion.

Perks & Benefits

This remote position is open to candidates based in Europe, promoting a flexible work environment. SignalWire emphasizes career growth through the development of innovative communication services and technologies. The company culture values collaboration and continuous learning, providing opportunities for engineers to enhance their skills in a dynamic setting.

⚠️ This job was posted over 3 months ago and may no longer be open. We recommend checking the company's site for the latest status.

Full Job Description

SignalWire is looking for an European-based WebRTC Software Engineer to help support our European expansion. SignalWire offers engineers the opportunity to build new services and systems critical to the success of a dynamic, fast-growing company-- to push forward the boundaries of communication and grow their career as SignalWire grows.

Please note: This role is only available to candidates who live in Europe and do not require sponsorship.

Job Duties:

  • Add new features to SignalWire’s CXML, RELAY, SWML, and Call Fabric products

  • Review peers’ designs and pull requests

  • Provide Tier 3 support for VoIP and WebRTC problems, joining customer calls when needed

  • Create unit and integration tests for CI/CD

  • Work with QA to ensure new features are testable

Qualifications:

  • Must be eligible to work within Europe without sponsorship.

  • Hands-on experience with C programming language

  • Experience extending C language VoIP/WebRTC applications like FreeSWITCH, Asterisk, Janus, Kamailio, OpenSIPS, rtpengine, or coturn

  • Familiarity with Docker and Git

  • Knowledge of WebRTC, SIP, SDP, RTP, SRTP, DTLS-SRTP, ICE, TURN, STUN, RFC2833/RFC4733, G.711

  • Hands-on experience with Typescript/JavaScript a bonus

  • Browser WebRTC API experience a bonus

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