OpenSIPS RTPEngine audio issue

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$30-250 USD

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Hi,

We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client.

Our flow of calls is like this:

WebRTC client -> OpenSIPS -> FreeSWITCH

The system is deployed on Azure.

We are looking for experienced person who has done such work and quickly help us.

VoIP Asterisk PBX 斐讯通信

项目ID： #35890818

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4个方案 远程项目 活跃的1 年前

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amelantoney的简历照片
amelantoney

This might be an issue with webrtc connectivity with freeswitch SIP handle. Please go through my past freeswitch and VoIP projects and customer feedback over ten years

$250USD 在1天里
(53条评论)
5.5
stylesiva的简历照片
stylesiva

I appreciate the Job Employment Invitation. I understand your requirement of Open SIPS RTP Engine audio issue. About VSOnline Services: We are a custom software development firm with 7+ years of extensive hands-on exp 更多

$135 USD 在7天内
(9条评论)
4.6
SyedRohaanAlam的简历照片
SyedRohaanAlam

Hello, I have more than seven years of experience in the office and more than three years of freelance experience in the required task and would like to help you with this task. Thanks for posting in my area of work.

$140 USD 在999天内
(0条评论)
0.0
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