[Rtcomm] [Rtcomm] [Bug 4212] Can make VoIP calls but cannot receive calls

Thu Mar 19 07:17:11 EET 2009

https://bugs.maemo.org/show_bug.cgi?id=4212

------- Comment #14 from maemo at subs.maneos.org  2009-03-19 07:17 GMT+3 -------
Mikhail: I may be opening a can of worms, but if this syslog output is correct:

------------------------------------------------------------------------
REGISTER sip:proxy01.sipphone.com SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.100:49152;rport;branch=z9hG4bK8QQrQQKNry0mc
Max-Forwards: 70 
From: <sip:17473312118 at proxy01.sipphone.com>;tag=8vr04mcHB438Q
To: <sip:17473312118 at proxy01.sipphone.com>
Call-ID: 62942b73-8e49-122c-4381-00194fd4ea6f
CSeq: 112567508 REGISTER
Contact: <sip:17473312118 at 192.168.1.100:49152;transport=udp>
User-Agent: Telepathy-SofiaSIP/0.5.4 sofia-sip/1.12.8
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, UPDATE
Supported: timer, 100rel, path
Content-Length: 0 
------------------------------------------------------------------------

it looks like STUN isn't working, at least on the SIP side.  Normally symmetric
response routing would take care of things, but in cases like this STUN might
make a difference (and probably explains why the gizmo UA works).  Is it a
known issue and/or worth filing a bug about it?

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