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SIP Jobs
SIP (Session Initiation Protocol) is an important industry standard protocol that is widely used for establishing, modifying and terminating multimedia sessions or voice and video calls across IP networks. It allows for a high degree of flexibility regarding codecs, and media types that can be interchanged during the session. A SIP Engineer is a skilled software developer/programmer that specializes in network collaboration protocols like SIP, RTSP and RTP that help enable peer-to-peer communication over the internet, reducing costs and allowing for more efficient deployment of voice and video conferencing.
A SIP engineer can do many things ranging from building Voice over IP systems to creating specialized multimedia applications. They will build sophisticated systems with various components such as audio encoders/decoders, media gateways, signaling gateways and user agent clients. They are also responsible for configuring and optimizing the system to achieve the best results and creating real-time simulations like teleconferencing or IVRs (interactive voice response).
Here’s some projects that our expert SIP Engineer made real:
- Secure upgrades for open source communications protocols
- Debugging and solving complex VoIP issues
- Carrier grade server implementations in cloud environments
- System wide port configuration optimization
- Integrating complex third party applications with existing softwares
- Coordinating German DIDs/Voip Numbers with PBXs
- Developing automated communications using Python
SIP Engineers are always in demand as great knowledge in this protocol is essential for setting up reliable communication services with low latency at an affordable rate. At Freelancer.com you can hire an experienced SIP Engineer who not only understands the inner workings of current standards but also able to keep up with new developments in the communication world. Post your project today and get the expertise you need to create a powerful yet cost effective communications environment!
From 2,629 reviews, clients rate our SIP Engineers 4.9 out of 5 stars.Hire SIP Engineers
SIP (Session Initiation Protocol) is an important industry standard protocol that is widely used for establishing, modifying and terminating multimedia sessions or voice and video calls across IP networks. It allows for a high degree of flexibility regarding codecs, and media types that can be interchanged during the session. A SIP Engineer is a skilled software developer/programmer that specializes in network collaboration protocols like SIP, RTSP and RTP that help enable peer-to-peer communication over the internet, reducing costs and allowing for more efficient deployment of voice and video conferencing.
A SIP engineer can do many things ranging from building Voice over IP systems to creating specialized multimedia applications. They will build sophisticated systems with various components such as audio encoders/decoders, media gateways, signaling gateways and user agent clients. They are also responsible for configuring and optimizing the system to achieve the best results and creating real-time simulations like teleconferencing or IVRs (interactive voice response).
Here’s some projects that our expert SIP Engineer made real:
- Secure upgrades for open source communications protocols
- Debugging and solving complex VoIP issues
- Carrier grade server implementations in cloud environments
- System wide port configuration optimization
- Integrating complex third party applications with existing softwares
- Coordinating German DIDs/Voip Numbers with PBXs
- Developing automated communications using Python
SIP Engineers are always in demand as great knowledge in this protocol is essential for setting up reliable communication services with low latency at an affordable rate. At Freelancer.com you can hire an experienced SIP Engineer who not only understands the inner workings of current standards but also able to keep up with new developments in the communication world. Post your project today and get the expertise you need to create a powerful yet cost effective communications environment!
From 2,629 reviews, clients rate our SIP Engineers 4.9 out of 5 stars.Hire SIP Engineers
Capture Live GSM Call Voice Packets
3 days left
I am looking for an experienced telecom/Android developer who can capture the live voice stream (two-way audio) of an active GSM (cellular) phone call. Requirements: Capture the live GSM call voice packets/audio stream. Access both incoming and outgoing audio during the call. Provide a technical approach for capturing the live voice stream. Explain any limitations or device-specific requirements. Share proof of concept or previous experience with similar work, if available. Please apply only if you have experience with GSM telephony, mobile voice processing, or low-level telecom/audio systems.
$17 / hr
Average bid
Vibe Optimizer & Gateway Configuration
2 days left
Verified
I have a fresh install of the Vibe server application running as a VoIP bandwidth optimizer and I’m stuck at the configuration stage. My immediate goal is to finish the initial setup, fine-tune the optimizer settings, and then bring a VoIP gateway online on a public IP so calls flow end-to-end without drops or quality loss. A screenshot of the current network layout is attached for reference. Here’s what I need: • Apply the right Vibe configuration parameters so it actually starts shaping traffic efficiently. • Map the public IP, open the correct SIP/RTP ports, and register the gateway so it’s reachable from outside the LAN. • Walk me through any CLI or web-GUI steps I’ll need to repeat later (firewall rules, NAT, codec or QoS tweaks). •...
$120
Average bid
Urgent 3CX SIP Trunk Setup
1 day left
I need a complete 3CX instance brought online for an Australian operation today—ideally within one to two hours of kickoff. The build must include a working SIP trunk (no provider pre-selected, so feel free to choose the quickest reliable option), inbound and outbound calling, and an IVR that plays my already-prepared pre-recorded messages. Beyond this first deployment, I regularly set up similar centres in the UK, AUS, and USA, so a smooth, repeatable process is important. Deliverables • Fresh 3CX install (cloud or on-prem, your choice) • SIP trunk fully registered and tested for Australian DIDs • IVR configured to play supplied audio prompts on both inbound and outbound flows • User extensions created as required and test calls recorded as proof of s...
$103
Average bid
Middleware SIP Development to Detect and Cut Voicemail Calls
2 days left
Verified
Desarrollo de Middleware SIP para Detección y Corte de Llamadas a Buzón de Voz Descripción del proyecto Busco un desarrollador con experiencia sólida en VoIP/SIP para desarrollar un middleware que permita detectar automáticamente cuando una llamada ha sido contestada por un buzón de voz y finalizarla de forma inmediata, minimizando costos de terminación. El middleware deberá funcionar tanto en escenarios donde el buzón responde **antes del 200 OK** (Early Media - respuestas 183 Session Progress con audio) como **después del 200 OK**, cuando la llamada ya fue contestada y comienza la facturación. Objetivos * Detectar buzones de voz durante Early Media y mediante análisis del audio de la llamada. * Final...
$2020
Average bid
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