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 основании 531 отзывов клиентов, рейтинг SIP Engineers составляет 4.98 из 5 звездочек.
Нанять SIP Engineers
I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.
This is a test task to find the right partner for onging work on this topic. Expected solution time: in a few weeks, we focus on quality-delivery & honest-estimation more than "quick & dirty" or "overseller" Your task is to help us to build the evil side ;-) windows (we provide linux) Ideally you have also mac experiences too In the case you have to build a dll of pjsip for windows, which detects incoming SIP/VoIP calls and take over tasks managed in a issue tracker for the java, c++ side and for linux, windows and mac Examples: ) So the app has to work on a desktop (windows, linux, macOS) and have to communicate with a SIP provider only. e.g (we will share you a fully working sip account after award) Later (not scope here) ports to iOS/Android required to...
we are looking for a exceptional good expert about SIP protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome t...
An Android tablet will be placed at the door. The tablet will log in to a local SIP server, set in the settings page. During idle state, it will show a static image or video of the user's choice, set in the setting's page. When the user taps on the screen, he will be presented with 2 choices, "Guest" or "Delivery" Tapping on 1 of the choice will initiate a SIP video call to the preset extension number. The extension number will be set in the settings page. Video of the conversation will be recorded. Video will only be from tablet to server, there will not be any video from server to the tablet. There will only be audio from server to tablet. After the call ends, it will return to idle screen. 1 setting field to define where the videos will be saved, opt...
I need some DID experts with a long experience who can set up the whole process which is needed to create DID numbers together with the Telecom company. Please do not respond if you do not have that experience.