[asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

[Mark Michelson]

On the reported issue, CSipSimple was a SIPS-using client that the 
reporter used. CSipSimple uses PJSua under the hood, so it may be common 
for PJSua-based clients (e.g. Blink) to use SIPS for secure calls.

I'm in a different environment today and I might be able to test with 
Blink myself.

On 01/15/2015 02:14 PM, Olle E. Johansson wrote:
>
> On 15 Jan 2015, at 21:07, Mark Michelson <reviewboard at asterisk.org 
> <mailto:reviewboard at asterisk.org>> wrote:
>
>> it feels like a bug that I can send a request to a SIPS URI over UDP and that Asterisk will accept the request.
> +1
>
> I can't remember any SIPS-using clients, can't say I've looked hard 
> though. Anyone that can help testing Mark's patch?
>
> /O :-)
>
>

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