Closed GoogleCodeExporter closed 9 years ago
I think this issue is related/same than issue 2776
Can you confirm?
Original comment by r3gis...@gmail.com
on 30 Nov 2014 at 11:34
Yes this is the same..
Original comment by ultrab...@gmail.com
on 30 Nov 2014 at 11:39
Can we add "transport=tls" in INVITE message in the cssipsimple options
Original comment by ultrab...@gmail.com
on 30 Nov 2014 at 11:45
Yes, it's done when you specify it in the sip uri you dial (switch to text
dialer mode and enter the full sip uri).
However, I recommand to set your sip server as your sip proxy and set transport
to tls in account configuration. In this case, no need to specify the transport
to reach the remote part since your SIP packet will go through your sip server
using the TLS transport as specified by your configuration.
That's actually the configuration most people need as they want to go through
their sip server to place the call (even in case the remote is not
hosted/registered on your server). Will grant at least that the path between
you and your server uses TLS.
If you do not specify the transport in proxy or in sip uri to call, the regular
SIP RFC specification applies and the call will use remote port 5060 and
transport UDP (the default in sip protocol).
Original comment by r3gis...@gmail.com
on 30 Nov 2014 at 11:59
I forgot to attach a full example that might be usefull to clarify :
sip:remote.user@his.sip.server:5061;transport=tls
Original comment by r3gis...@gmail.com
on 30 Nov 2014 at 12:00
Thank you it's working fine ...
Original comment by ultrab...@gmail.com
on 30 Nov 2014 at 1:04
Original issue reported on code.google.com by
ultrab...@gmail.com
on 29 Nov 2014 at 11:04