[Freeswitch-users] Transporting SIP over TCP

[bakko]

I'm just tryng TCP now and work (latest trunk)

I done this test with eyebeam phone registeres to TCP port and call other ip 
phone (registered over UDB port).

The audio work fine in both directions.

My configuration:

FS --> internet --> NAT --> Phones

In the internal profile:

Call-ID:        OTIyNzRlNTQxNTk5NzBhYzMzYzM1YmQyNTgzNTA3MTk.
User:           1004 at nydomain.com
Contact:        "User" 
<sip:1004 at 87.12.101.59:13035;transport=TCP;rinstance=f3fd252ea20dd34a>
Agent:          eyeBeam release 1102u stamp 52345
Status:         Registered(TCP)(unknown) EXP(2009-08-16 08:59:38)
Host:           mydomain.com
IP:             XXX:XXX.XXX.XXX
Port:           13035
Auth-User:      1004
Auth-Realm:     mydomain.com