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Voip Software Jobs

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Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Hello, I want a script to use for my Raspberry pi, for voice morphing. The requests are: - Apply effects in real time to an audio input (external microphone - external sound card); - the effects can be customize in the script. - if there is a way to change the effects by a 3x4 keyboard. - I need to be a stand alone solution, plug and play. (after customization); I need it right away. Thank you
Skills: VOIP Software Circuit Design Embedded Linux Python
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
This is a MS Voice test bed for SIP trunk compatibility testing to allow Skype For Business to make and receive calls via external VoIP provider. It must allow - local Skype for Business client registration - SIP Invites will be sent to provider's external SIP proxy to initiate a call in E.164 format i.e. SIP/4420812345678@proxy_hostname - Lync requires TLS+SRTP, sip provider only supports TLS - mode change required - receive inbound calls to active SIP proxy registration - our understanding is that we only need two servers: one with DC, another one with front end and mediation server roles colocated, feel free to correct this. No messaging, conferencing, video services are required. We merely need to have a working prototype that will allow us to connect a Lync install to an external VoIP provider and use their SIP trunk for placing and receiving calls. We will also need this to be accompanied by a short setup guide with screen grabs illustrating this setup. Not the entire install, only the SIP trunk provider hookup part. You will be provided with required number of Win2012 servers with remote access to fit your requirements. Alternatively you can spin those VMs on any VPS provider you like, costs will be reimbursed.
Skills: VOIP Software Microsoft Lync Server Microsoft Server SIP
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We're looking to have SMS/MMS t.38 faxing and Conference Bridge Apps made. SMS/MMS would need to send SMS replies back through email and possibly even start from an email if possible. Would be nice also to have a conference bridge as well with some basic feature sets. t.38 faxing could work with either Plivo or Voip Innovations.
Skills: VOIP Software API Development iPhone App Development PHP
Hourly - Intermediate ($$) - Est. Time: More than 6 months, 10-30 hrs/week - Posted
Hello, we have already installed 3cx free version on our machine. Right now we have problem to configure Inbound and Outbound route for our system. We don't have SIP Provider directly, but we want to use our Router Gateway for the calls. Because our ISP provider let us make call voip and the configuration is on the router directly. We can access the configuration information for voip call. Please see the attached document. For further question please get in contact with me.
Skills: VOIP Software SIP VOIP Administration Windows Administration
Hourly - Entry Level ($) - Est. Time: 3 to 6 months, 30+ hrs/week - Posted
Responsibilities: • Design and build new code, tools and applications related to videophone initiatives • Translate business requirements into logical, component-based technical designs • Provide ongoing code-level maintenance and support for videophone applications • Drive resolution of all issues that come up during development phase • Clearly communicate project status to key stakeholders throughout entire development process • Create comprehensive technical system documentation and provide handoff training to business and other development staff • Provide valuable input for technical meetings in regards of all aspects of VoIP development Qualifications: • Minimum 5+ years VoIP development experience, with proven ability to design mobile and desktop applications for iOS platform. • Requires 5+ years experience with SIP, RTP, TCP, TLS, UDP protocols • Requires 5+ years experience developing audio and video codecs used in VoIP, such as G.711, G.722, OPUS, H.264 and VP8 • Requires 5+ years experience developing and improving jitter buffer techniques, circular buffers, packet loss compensation, redundancy compensation, media device usage for Android, Apple iOS, OS X and Windows platforms • Experience building mobile and desktop applications for Android, Apple iOS, OS X and Windows platforms • Experience writing and debugging Android Java code using AndroidStudio IDE • Experience writing and debugging Apple iOS code using Xcode IDE • Experience writing and debugging Windows Desktop code using DevStudio IDE • Experience in using Linux shell commands to administer a remote server • Experience of understanding SIP, XMPP and RTP log files and live output • Strong understanding of the development process within a company making VoIP applications • Knowledge and proven ability to use source control systems and management toolssuch as Github, JIRA and Confluence. • Experience juggling multiple tasks, communicating to team leaders and meeting strict deadlines on development projects • Demonstrated organizational skills & ability to multi-task in a fast-paced environment with competing priorities. A generic NDA with contractor agreement is to be signed ahead of releasing access to git.
Skills: VOIP Software Apple Xcode VOIP Administration
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, 10-30 hrs/week - Posted
I'm working with a radio company partner Icom, and we have purchased this SDK/API for integrating into there communication architecture. Using the SDK we can control messaging, radio id, and audio in/out. What i'm interesting in is developing a custom audio gateway plugin that would allow audio to flow in and out to SIP enabled topologies (like asterisk/Etherstack and many others) . We have the API for this. Normally I ask them to work on a transcoder, but the audio codec ICOM uses is proprietary and they only licenced us to use a hardware decoder/encoder (API's provided). The AUDIO IN/OUT and Push-To-Talk and Incoming Voice Call is exposed in the API layer, which may allow use to code to an open SIP stack. I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). Note the current development will be in windows, but possibly moved into windows embedded later.
Skills: VOIP Software Asterisk Microsoft Visual Studio OpenSIPS
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