>* Is the SIP signaling still going fine via websocket? Do you get callee
*>* ringing?
*>>* Cheers,
*>* Daniel*
Hello Daniel
Actually SIP signalling goes perfect, and i can see in RTP asterisk
log that RTP packets are being sent, so I suppose there is something
stuck somewhere in
Hi,
If you are using rtpengine for the rtp you might have to upgrade to a
newer version.
On my system crypto negotiation in rtpengine started failing after a
openssl update in January. Recompiling a newer version from git fixed that.
regards
M
On 02/13/2015 09:56 AM, Rahul MathuR wrote:
Hello Manuel,
To support the hypothesis of crypt libs screwing the logic, you can try a
'secure call' without using webrtc.
If them are to be blamed; your 'secure call' won't be successful.
Aside this, you can get a better idea of what has dwell-ed behind the
curtains by looking at syslogs.
Hop
On 12/02/15 17:04, Manuel Camargo Lominchar wrote:
> This might be a weird question
> I've been operating for some months with my kam + asterisk + webrtc
> sipml5 based system inside an Ubuntu Server
>
> Today I had the idea to apt-get upgrade my system and... now the whole
> system webrtc communi
This might be a weird question
I've been operating for some months with my kam + asterisk + webrtc sipml5
based system inside an Ubuntu Server
Today I had the idea to apt-get upgrade my system and... now the whole
system webrtc communication service is dead
Here is the VAST list of packages updat
