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!

Nga 2,153 vlerësimet, klientët vlerësojnë SIP Engineers 4.93 nga 5 yje.
Punësoni 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!

Nga 2,153 vlerësimet, klientët vlerësojnë SIP Engineers 4.93 nga 5 yje.
Punësoni SIP Engineers

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    VICIdial Cloud Installation & 32-SIM Gateway Integration Expert Needed Job Description: We are currently using a 32-SIM GSM Gateway setup with VICIdial for inbound and outbound calling campaigns. We are looking for an experienced expert to migrate, configure, and optimize our system on a cloud-based environment. Scope of Work: Install and configure VICIdial on a cloud server (AWS / DigitalOcean / GCP) Integrate existing 32-SIM GSM Gateway with the cloud setup Configure SIP / VoIP connectivity Set up and optimize: Outbound campaigns Inbound call handling Ensure: Call quality Call routing System stability Enable: Call recording Reporting & analytics Fix data posting / call log sync issues (if any) Provide basic training and post-setup support Key Challenges to Address: Cloud data...

    €22 Average bid
    €22 Oferta mesatare
    3 ofertat

    Vicidial is already live on our cloud server; what I need now is an on-site engineer in Hyderabad who can bridge it to a 32-port Dinstar GSM gateway sitting in our office. Your task is strictly configuration and testing—no fresh installs. Scope of work • Register the Dinstar gateway as a SIP trunk inside Vicidial, keeping a close eye on call-quality parameters and full compatibility with the gateway. • Analyse the current router, decide which ports must be opened, then set up forwarding so traffic reaches the cloud server cleanly. No predefined template exists, so you will determine the rules and apply them. • Run end-to-end test calls, troubleshooting the single audio problem we’ve seen so far: distorted sound. I expect clear audio on both legs before sign-off....

    €20 Average bid
    €20 Oferta mesatare
    2 ofertat

    ## 1. Project Scope The service provider will design, install, and configure a production-ready VoIP Call Center system using Asterisk on a Linux-based environment. The scope includes: * Server Environment: Installation and hardening of the latest stable Asterisk version on a cloud-based or local Linux server (Ubuntu/Debian). * Trunking & Connectivity: Integration of [X] number of SIP Trunks for external inbound and outbound calling. * Extension Management: Creation and configuration of [X] agent extensions supporting PJSIP/SIP protocols. * Call Flow Logic (Dialplan): * Development of a multi-level IVR (Auto-Attendant) menu. * Implementation of time-based routing (Business Hours vs. After Hours). * Queue Management (ACD): Setup of call queues with specific distribution strategies (...

    €3512 Average bid
    €3512 Oferta mesatare
    38 ofertat
    3CX Professional Licensing Partner Needed
    2 ditë left
    Të verifikuara

    I am looking for a 3CX Professional licence partner and need a certified partner who can both supply the key and stay available for the long term. First, I want your help choosing the correct user-count edition, issuing the licence, and making sure activation goes through smoothly in my current SIP-trunk environment. Once the system is live I expect to lean on you whenever technical troubleshooting is required and to keep the platform patched and secure with regular updates and maintenance. Please outline your current 3CX partner status, typical turnaround for issuing new Professional licences, and how you normally handle remote support requests. If you already provide monitoring or update plans, mention those as well so I can see how they would fit my needs.

    €99 Average bid
    €99 Oferta mesatare
    3 ofertat

    A FusionPBX (FreeSWITCH) installation and custom call flow configuration will be performed on a cloud server (Debian-based). The technical requirements of the project are as follows: 1. Server and Panel Installation: Installation of an up-to-date and stable FusionPBX on a Debian operating system. Configure firewall and Fail2Ban settings. 2. SIP Trunk and Incoming Call Management: Definition of SIP Trunk information (e.g., Telnyx / Verimor) provided by me. Creation of an enterprise welcome menu (IVR). Directing callers to the relevant extensions via the voice menu. 3. Dynamic Mobile Phone Routing (Critical Point): Incoming calls to extensions will be routed directly to their mobile phones. Important: The mobile numbers to which extensions are routed will not be fixed. Each extensio...

    €92 Average bid
    €92 Oferta mesatare
    76 ofertat

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