i have configured a server with Asterisk 12.1.1 with two extensions 101 and 102 and enabled transport ws in sip.conf.
In the same server i have installed sipml5.
The server has ip address: 192.168.1.10
Ext 101 (from my pc - simpl5): 192.168.1.1
Ext 102 (from another pc - simpl5): 192.168.1.2
So all the devices are in the same network.
When i call 102 from ext 101 i get this error: Got SIP response 603 “Failed to get local SDP” back from 192.168.1.2:56282
When i call 101 from ext 102 i get this error: Got SIP response 603 “Failed to get local SDP” back from 192.168.1.1:56285
Why this happens? I can’t understand what happens…
Thanks and regards,
Show us the complete sip debug.
The phones are refusing to talk to you. 603 is “call declined”. It is basically the same as 403, but more definitive.
i have followed this “guide”: viewtopic.php?f=1&t=90167
Moreover, i added this option: nat=force_rport,comedia and audio started working.
Now, i have some problems with video: when i try to do a video call, Asterisk generates this error: Rejecting secure video stream without encryption details: video 59831 RTP/SAVPF 100 116 117
What does it mean? How can i solve it?
Thanks and regards,
It means that the caller requested encrypted video but didn’t say how to encrypt/decrypt it.
The normal solution to this sort of incompatibility is to disable encryption, but I don’t know if you can do that with WebRTC.
You would need to provide the information that has already been requested in order to be more precise and/or determine if Asterisk is at fault.
in webRTC, encryption is mandatory, i’m having the same issue for weeks and i can’t find a good solution,webrtc2sip is a Doubango gateway, but introduces a 5s delay in the conversation, this is an unacceptable delay for my project, but maybe suits good for your purpose.Now i’m trying to use Kamailio as SIP server.
You shouldn’t use a development API like webrtc for production stuff. There is no RFC yet and in the nearest future things will going to break down with the recent changes in Chrome.
In the meanime you can provide logs, configs and debugs output for your issue in order to get help and with that others can reproduce your enviroment and maybe file a bug to the Asterisk or the WebRTC API.
I’m using webRTC because it’s mandatory in the project i’m working in. I’ve read that nowadays video calling through SIP with Asterisk isn’t supported. Is this feature in Asterisk’s roadmap?
Thank you in advance,
did solve this problem?? can you post for us your file configuration?