[SR-Users] websocket transport protocol in Record-Route header

[Daniel-Constantin Mierla]

On 27.09.20 15:43, Juha Heinanen wrote:
> Daniel-Constantin Mierla writes:
>
>> For SIP URI the parameter value is always ws:
>>
>>   * https://tools.ietf.org/html/rfc7118#section-5.2
> Yes, according to that RFC, but does it make any sense, since how you
> tell based on usrloc received value if ws or wss was used for the
> registration?

I never went to read all the specs and discover the rationale of having
only transport=ws for URI, but I remember that initially the websocket
code in Kamailio (iirc, was developed mainly by Peter Dunkley) was using
wss, but then had to be updated to match the rfc.

I think that from standardisation point of view, the websocket can be
only via tls/https, the version over tcp was more like for
troubleshooting. On the other hand, the Via header can have WS and WSS
(afaik from RFC). From the point of view of a server that only accepts
websocket connection, never opens, practically it has to find an active
connection that matches on remote peer ip and port, with websocket
protocol flag set (it is impossible to have two matching the same, one
over tcp and one over tls).

If nobody else here can add more, this aspect can be clarified by asking
on sip-implementors mailing list or maybe ietf sipcore wg mailing list.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
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