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webrtc - No have audio/video in freeswitch - Stack Overflow

https://stackoverflow.com/questions/33520158/no-have-audio-video-in-freeswitch

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No audio with FreeSWITCH webrtc · Issue #119 · …

https://github.com/versatica/JsSIP/issues/119

No audio with FreeSWITCH webrtc #119. Closed frisb opened this issue Jun 27, 2013 · 45 comments Closed No audio with FreeSWITCH webrtc #119. frisb opened this issue Jun 27, 2013 · 45 comments Comments. Assignees No one assigned Labels None yet Milestone No milestone Linked pull requests

WebRTC/SipJS No Incoming Audio | FusionPBX Forums

https://www.pbxforums.com/threads/webrtc-sipjs-no-incoming-audio.4097/

Apr 19, 2020. #1. I have a browser-based WebRTC softphone setup using SipJS and FusionPBX. My FreeSwitch version is 1.10.2. If I call out from SipJS to an internal extension or an external pstn number, audio is working both ways. If I call in to SipJS from a SIP softphone or and external pstn call then the SipJS client cannot hear anything (but ...

sip - No audio in WebRTC and Asterisk - Stack Overflow

https://stackoverflow.com/questions/28501309/no-audio-in-webrtc-and-asterisk

All WebRTC clients are inside local network, so ICE isn't needed here. But the bad way - ICE is the problem here. I've made a lot of tests and found that if call initiator is Web RTC client and there is some delay in answer (10-25 seconds) - audio is completely absent.

Configuring FreeSWITCH for WebRTC | Packt Hub

https://hub.packtpub.com/configuring-freeswitch-webrtc/

In the article written by Giovanni Maruzzelli, author of FreeSWITCH 1.6 Cookbook, we learn how WebRTC is all about security and encryption. Theye are not an afterthought. They’re intimately interwoven at the design level and are mandatory. For example, you cannot stream audio or video clearly (without encryption) via WebRTC.

FreeSwitch + Mode Verto + Webrtc + Android + unable to ...

https://stackoverflow.com/questions/69435395/freeswitch-mode-verto-webrtc-android-unable-to-make-call-from-android

Show activity on this post. and FreeSwitch but only got success to make uni-directional communication (SIP phone to an android client, voice communication in both devices successfully). But when I try to make a call from an android client to a SIP phone it doesn't work. Steps: From Web To mode-verto android Client (Working scenario): Login Call:

FreeSWITCH And The Opus Audio Codec - FreeSWITCH - …

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec

About. This page derives from a document written initially by Dragos and Giacomo in October 2016 (you can find the original attached here for historical reasons). It is an attempt to provide a general view of the interaction between FreeSWITCH and Opus, under several points of view: use, configuration, development, debug, installation.

Setting up FreeSWITCH WebRTC functionality – dOpenSource

https://dopensource.com/2017/01/21/setting-up-freeswitch-webrtc-functionality/

This tutorial will go over how to setup WebRTC on FreeSWITCH using a certificate from letsencrypt. WebRTC is a protocol which allows voip calls to be conducted over a web browser without additional plugins or software. This tutorial will assume you are Debian 8, which is the recommended OS for production FreeSWTICH servers.

Integration Of FreeSWITCH Into WebRTC – What Do You Need ...

https://www.moontechnolabs.com/blog/webrtc-and-freeswitch-what-this-combination-means/

2) WebRTC is on the way. 3) The working process. 4) Security or encryption. 5) More than peer-to-peer communication. 6) Gateways and servers. 7) FreeSWITCH is a combo of all. 8) Final words. WebRTC has been around for a long time, and you probably know a thing or two about it already.

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