[jitsi-issues] #1300: Dialing on a SIP registrarless account always uses UDP

Jitsi Trac trac at jitsi.org
Sat Aug 9 13:13:53 CEST 2014


#1300: Dialing on a SIP registrarless account always uses UDP
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 Reporter:  pocock       |       Owner:  somebody
     Type:  defect       |      Status:  new
 Priority:  8            |   Milestone:
Component:  development  |  Resolution:
 Keywords:               |
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Comment (by pocock):

 Emil, that is only true if the destination is an IP address and there is
 no transport parameter:

 http://tools.ietf.org/html/rfc3263#section-4.1

 If the destination is a domain, then the way to proceed is:

 a) check for a transport parameter in the request URI, e.g.
 sip:bob at example.org?transport=tls
      (Jitsi currently ignores that)

 b) check for a NAPTR record

 Also notice the comment under the NAPTR stuff in RFC 3263 "A client
 resolving a SIP URI SHOULD retain records with "SIPS" as the protocol, if
 the client supports TLS."

 Do you think there is anything in the stack that will make this difficult
 for Registrarless accounts?  I also ran into trouble trying to encourage a
 normal account to make outgoing calls without being registered:
 https://github.com/jitsi/jitsi/pull/51

-- 
Ticket URL: <https://trac.jitsi.org/ticket/1300#comment:2>
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