Sip Jobs

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Fixed-Price - Expert ($$$) - Est. Budget: $1,000 - Posted
Looking for an expert to compile siphon ( or a sip client that really works with either kamailio + rtp proxy or freeswitch or asterisk ) with our company logo and skin.
Skills: SIP
Fixed-Price - Expert ($$$) - Est. Budget: $1,000 - Posted
Hi I have a high throughput and high load C program that is an open source project. I need help with modifying its internal data structure so it can be fully optimised, run in parallel, and utilize all cores of the server. You will need to be able to use Linux memory usage, gcc tuning tools, to find out the bottle neck and fix it in the code.
Skills: SIP C
Fixed-Price - Intermediate ($$) - Est. Budget: $300 - Posted
(Gateway between WebRTC client and switch/PSTN Gateway) When PSTN call is made from WebRTC client, it will connect to a WebRTC server for signalling. Further WebRTC server will send appropriate SIP signals to Asterisk using WS protocol. We will integrate JSSIP with WebRTC server so that proper sip signals can be generated and sent to Asterisk system. ... We will integrate JSSIP with WebRTC server so that proper sip signals can be generated and sent to Asterisk system. ... BB) Asterisk would need to transcode the opus codec to G729 No billing is required in WebRTC gateway. If required by asterisk, the sip ID and password will be available in database CC) incoming calls from switch or PSTN Gateway will be sent to WebRTC client.
Skills: SIP Asterisk WebRTC
Fixed-Price - Expert ($$$) - Est. Budget: $2,000 - Posted
For this feature we need a consultant who has experience in successfully setting up SIP or WebRTC based VOIP systems. The consultant also needs to guide the iOS and the Android development team on the client libraries to use to connect to the server (like SIPDroid C-SIP in Android and PJSIP in iOS or LinPhone on both platforms) The VOIP feature should work in all countries and between all combination of networks like the below (see PDF attached). ... The consultant also needs to guide the iOS and the Android development team on the client libraries to use to connect to the server (like SIPDroid C-SIP in Android and PJSIP in iOS or LinPhone on both platforms) The VOIP feature should work in all countries and between all combination of networks like the below (see PDF attached).
Skills: SIP VOIP Administration VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
We are Looking to setup our own virtual PBX based on one of the open source (such as asterisk) We are looking to have our PBX where client can call a number and get auto attendent and various of other features like call hold, music on hold, transfer call, call merge etc. The system will work the following way: Caller > auto attendent > #ext > forward to a 3rd party number.
Skills: SIP Asterisk VOIP Administration VOIP Software