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!

На підставі 2,874 відгуків клієнтів, рейтинг SIP Engineers становить 4.91 із 5 зірочок.
Найняти 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!

На підставі 2,874 відгуків клієнтів, рейтинг SIP Engineers становить 4.91 із 5 зірочок.
Найняти SIP Engineers

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    *** STRICT RULES - READ BEFORE BIDDING *** 1. DO NOT BID if you do not have direct, hands-on experience with ALL of these specific devices: - Cudy P2 5G Router - DrayTek Vigor 2915 Router - Yeastar S20 PBX I will NOT explain how these devices work or provide any tutorials. 2. ALL work MUST be done remotely via AnyDesk ONLY. No direct access, VPN credentials, or passwords will be handed over. 3. Do NOT message me asking "What is the problem?" or "What is your budget?". The entire problem, infrastructure, and scope of work are detailed precisely below. Your bid is your final price. Any messages asking these questions will result in an immediate BLOCK. --------------------------------------------------------- --- PROJECT DETAILS & SCOPE OF WORK --- We are looking for...

    $131 Average bid
    $131 Сер. заявка
    20 заявки
    Low-KYC UAE High-Volume SIP Trunk
    2 дні(-в) left
    Верифіковано

    My operation needs a dependable SIP trunk that can originate and terminate voice traffic to United Arab Emirates destinations while keeping KYC requirements to the absolute minimum—ideally none. I am open to unlicensed wholesale routes as long as they deliver consistent quality and uptime. The trunk must comfortably handle tens of thousands of voice minutes per month, with enough concurrent channels to prevent congestion. Competitive per-minute rates, clean CLI presentation, and stable routing are critical. Please indicate whether you offer SIP registration or IP authentication, the codecs you support (G.711, G.729, etc.), and any fair-use or traffic restrictions. Deliverables • Active SIP credentials or IPs ready for testing • Current rate sheet and payment terms &bu...

    $50 Average bid
    $50 Сер. заявка
    1 заявки
    Android-iOS VOIP Voice Call App
    2 дні(-в) left
    Верифіковано

    We are currently working on our mobile application, GCM Dialer, which handles both SIM and SIP calls. We have successfully implemented the conference screen for the SIM and SIP merge option, but we are encountering an issue where voice is not being exchanged between all parties. Below is the scenario we are testing: - User-1 has the GCM Dialer app installed and set as the default dialer. - User-2 calls User-1 via SIM, and they can exchange voice successfully. - User-3 then calls User-1 via SIP. User-1 answers, which puts User-2 on hold. User-3 and User-1 can exchange voice successfully. - User-1 clicks the "Merge" button, and all three parties are brought into one call screen. The issue is that while User-1 can hear and speak to both User-3 and User-2, User-2 and User-3 cannot...

    $148 Average bid
    $148 Сер. заявка
    36 заявки
    Android UAE Local Calling Setup
    21 годин(-и) left
    Верифіковано

    I need a working UAE local number that lets me place and receive calls from my Android phone exactly as if I were inside the country. Linkus is the reference point, yet I’m open to any SIP softphone or cloud-PBX solution that delivers equal or better results. Voice quality must be stable on both Wi-Fi and 4G, and the same local caller ID has to appear to the person I’m calling. SMS integration is essential—I must be able to send and receive texts on the very same UAE number through the chosen app or an associated API. Acceptable solutions may involve: • Provisioning a UAE DID, SIP trunk or virtual number. • Full configuration on an Android softphone (e.g., Linkus, Zoiper, Grandstream Wave, etc.). • Demonstrated test call and SMS exchange showing local...

    $127 Average bid
    $127 Сер. заявка
    32 заявки

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