I have started to play around with webRtc. My first work is to implement websocket support in our existing SIP server of our company. Which is going public right now as named as SIP.ANYFIREWALL.COM. One of our company's webrtc support is allready running on cloud for TURN/STUN or as Relay server. This can be found on TURN.ANYFIREWALL.COM .
So I have implemented the Web SIP support (WebSocket with SIP protocol support) and seen some interesting output when I tested it sipml5.org 's webrtc client. I feel pleasure to say that it's working fine when I tested some Video call.
So feel free to test your websocket SIP client with our Web SIP server when it goes public. :)
Our Test Result:
Subscribe to:
Post Comments (Atom)
How to Generate and use the ssh key on Gerrit, github.io, gitlab, and bitbucket.
Details can be found here -
-
There is none who can replace him.At least the standard which he create in is life time in the running literature it never be replaceable.Th...
-
Server #include <sys/types.h> #include <sys/socket.h> #include <netinet/in.h> #include <arpa/inet.h> #include <...
-
ISSUE: Asterik need SQLITE3 , when it doesn't find this then shows the following warnings - configure: WARNING: *** Asterisk now uses...
Thanks for sharing your honest experience. When I first took a look at my head shots,
ReplyDeleteI wasn’t too thrilled with mine but you’ve given me a new perspective!
Virtual Edge