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Hi Max, Thank you for the response. I have captured the pcap as mentioned above. I have sent the logs to logs@flashphoner.com. Thanks for the help..:)

Hi Max, Thank you for your response. There still seems to be an issue with the SIP-WEB call with the Demo App launched. The media seems to be cut on both ends. Can you please help me this? Seems that the ICE is failing again. It would be really helpful if any insight is given on this..:)

Thank you for the response. As mentioned, I changed the rtp.conf as follows. [general] rtpstart=30000 rtpend=32000 ;icesupport=yes ;stunaddr=stun.l.google.com:19302 It seems that the same problem still persists. I am using Bria as the SIP endpoint and PhoneJS as the web user. The call gets...

SIP-WEB call seems to be failing for audio stream on both web and SIP endpoints but the call is getting established. The Chrome and Firefox seem to have the same behavior with respect to the media between the SIP and the WEB. No media on both ends. I am able to see that ICE is failing. Attached...

Hi Max, Thank you for the response. I wanted to understand the ways in which the presence of a user can be updated when he receives a call. I can probably give scenarios and explain the same. Scenario 1:- Web User is Available before the call. He goes to the busy status when he is on a call(Now...

Thank you for the response. Is there a support for the WCS server to integrate with any other XMPP client- server application?If so any documentation for the same? Can you also give an insight on the presence updation for a user who goes on a call(on hook and off hook) Thanks..:)

Hi Max, Thank you for your response. The audio calls are working fine. I got the required information about the working of ICE and DTLS with FlashPhoner. Could you please let me know the Video Codec support that FlashPhoner has and the XMPP support it has with Asterisk(or FlashPhoner App to be...

Thank you for your response. I had a query regarding disabling of ICE and DTLS. I am sort of confused since DTLS and ICE marks the basic functionality of a WebRTC communcation, with neither of them that is without ICE gathering and DTLS being absent, can the secure communication be...

Hi Max, I thank you for your response. It seems pretty much that the first 2 steps are covered. I am facing issues when I test the Demo app from my WCS server. I see that the offer and answer sent from the server does not have the DTLS/SRTP fingerprint in the SDP. From the asterisk logs I can...

I am trying a webrtc-sip via Asterisk call with Asterisk 14 and WCS Server version FlashphonerWebCallServer-5.0.2159. I am using the demo application for WebRTC(Phone and Phone Video) and Bria for SIP End Point. I am getting the following issue in the console of Asterisk [Apr 5 15:36:51]...

I am trying a webrtc-sip via Asterisk call with Asterisk 14 and WCS Server version FlashphonerWebCallServer-5.0.2159. I am using the demo application for WebRTC(Phone and Phone Video) and Bria for SIP End Point. I am getting the following issue in the console of Asterisk [Apr 5 15:36:51]...