[asterisk-dev] uncached realtime sip friends cannot set transport-protocol in sipregs

[Walter Doekes]

>> However, when asterisk calls the user, it always uses the "preferred"
>> transport, namely the first option: UDP, *even* though the user
>> registered with transport=tcp in the Contact header.
> Was it in the URI or in the header parameter? Can you please copy the
> contact here. If it's in the URI, it should be stored in the reg database.

Contact from REGISTER:

   Contact: <sip:200 at 127.0.1.1:1234;transport=TCP>

>> The cause of this simple: the sipregs table has no place to store the
>> transport -- except for the fullcontact column, see below -- ergo, it
>> does not know that the client prefers (and in many cases, *only* does)
>> TCP.

> It should at least be stored in the peer structure, if not cached to
> astdb and/or realtime.

The peer structure is destroyed automatically when not needed, right? 
Hence the rtcachefriends=no.

As my report below shows:
- cached peer structure does store the transport
- astdb does store fullcontact (but doesn't use it)
- realtime (sipregs) doesn't store fullcontact

> I can't comment on the TCP/TLS part of the code since that's been having
> a lot of issues (it's still marked experimental for a reason). If we go
> ahead and change the astdb structure, we should add
> - expiry time for the registration
> - asterisk version (to avoid issues when upgrading and make it possible
> to change the data structure in the future).

If possible, I don't want to touch the astdb at all. See below.

========================================================================

Tests done below with Asterisk PBX SVN-branch-1.8-r334453M
(with patches of ASTERISK-18354, ASTERISK-18355 and ASTERISK-18356, but
they are irrelevant for the outcome)

sip.conf:

[general]
allow=all
allowguest=no
nat=no
tcpenable=yes
transport=udp,tcp

========================================================================
rtcachefriends=yes
========================================================================

<--- SIP read from TCP:127.0.0.1:51806 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/TCP 127.0.1.1:1234;branch=z9hG4bK-9155-1-0
From: <sip:200 at 127.0.0.1:5060>;tag=9155SIPpTag001
To: <sip:200 at 127.0.0.1:5060>
Contact: <sip:200 at 127.0.1.1:1234;transport=TCP>
Expires: 600
Call-ID: 1-9155 at 127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0

*CLI> sip show peer 200
...
   ToHost       :
   Addr->IP     : 127.0.1.1:1234
   Defaddr->IP  : (null)
   Prim.Transp. : TCP
   Allowed.Trsp : TCP,UDP
   Def. Username: 200

*CLI> channel originate SIP/200 application Answer 1000
Reliably Transmitting (no NAT) to 127.0.1.1:1234:
INVITE sip:200 at 127.0.1.1:1234;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 127.0.1.1:5060;branch=z9hG4bK225249ad
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as24c20e82
To: <sip:200 at 127.0.1.1:1234;transport=TCP>
Contact: <sip:Anonymous at 127.0.1.1:5060;transport=TCP>
...

========================================================================
rtcachefriends=no
========================================================================

<--- SIP read from TCP:127.0.0.1:56794 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/TCP 127.0.1.1:1234;branch=z9hG4bK-9218-1-0
From: <sip:200 at 127.0.0.1:5060>;tag=9218SIPpTag001
To: <sip:200 at 127.0.0.1:5060>
Contact: <sip:200 at 127.0.1.1:1234;transport=TCP>
Expires: 600
Call-ID: 1-9218 at 127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0

*CLI> sip show peer 200
Peer 200 not found.

*CLI> sip show peer 200 load
...
   ToHost       :
   Addr->IP     : 127.0.1.1:1234
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : TCP,UDP
   Def. Username: 200

*CLI> channel originate SIP/200 application Answer 1000
Reliably Transmitting (no NAT) to 127.0.1.1:1234:
INVITE sip:200 at 127.0.1.1:1234;transport=TCP SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK20bd6397
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as4c5be1c5
To: <sip:200 at 127.0.1.1:1234;transport=TCP>
Contact: <sip:Anonymous at 127.0.1.1:5060>
...

========================================================================

Observe that we see "sip:200 at 127.0.1.1:1234;transport=TCP" in the R-URI.
But that's completely useless, as it's attempting to connect over UDP.

If the 'fullcontact' column is set in sipregs, which is *only* written
rtcachefriends=yes, this value is *read* from the db. If it is
empty it uses the value stored from the astdb, if available.

Your question did reveal that the behaviour is different on my
production system than on a clean install. But it's broken in both
cases.

In my case, I disabled the writing to astdb, so I don't get the nice
R-URI with transport=TCP.
https://issues.asterisk.org/jira/browse/ASTERISK-17294

But that's logical. If it's going to store the fullcontact, it better
well do it in the database and not in the astdb of the UA that the
REGISTER request happened to land on.

In either case, the transport from the R-URI is *not* used to determine
transport protocol.

========================================================================

Regards,
Walter Doekes
OSSO B.V.