webrtc
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CloudSIP – WebRTC SIP Phone & Browser Extension for Asterisk
Hello everyone, I'd like to share an open-source project I've been working on for the Asterisk community. CloudSIP is a WebRTC SIP Phone that allows users to make and receive SIP calls directly from a web browser without installing a traditional softphone. It is available both as a... -
Anyone exploring testing Verto/WebRTC with FreeSWITCH?
Hi All. Some time ago, I was finding about testing webrtc with verto using freeswitch, and found this project: https://sourceforge.net/projects/embedded-webrtc/files/Embedded-WebRTC/ It has normal public access and also under Wireguard VPN May some one interested on this? Regards.- fmateo05
- Thread
- freeswitch verto voip webrtc
- Replies: 0
- Forum: Open Discussion
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TUTORIAL Implement WebRTC in Asterisk
As part of contributing to the Asterisk community, VitalPBX has just published a new Blog on how to implement WebRTC with PJSIP in Asterisk from scratch. We invite you to read our new Blog which is in English and Spanish. English Version...- Rodrigo Cuadra
- Thread
- asterisk vitalpbx webrtc
- Replies: 4
- Forum: Open Discussion
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Trying to set caller id using WebRTC and jssip
Hi, Im trying to use caller id with Asterisk and JSSip I can make calls and everything works as it should, except caller id only showing unknown. I’ve tried to Dynamic External CID to true which changes the caller ID from my sip trunks DID to unknown. I can see in my trunks log, that...
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