Archived from groups: comp.dcom.voice-over-ip (
More info?)
Rob Walford wrote:
> thanks for that.
> ive had a go, but still no joy.
>
> im not getting authentication failure message now, but i see four of
> these:
>
> SIP/SDP Request: INVITE sip:10000@217.10.79.219:5060, with session
> description
The "INVITE" message is the one sent by the caller to the called party,
usually passing through a middleman (the "outbound proxy") belonging to
the provider. In cases like yours the message is also sent to the
provider's server because the device you are calling is supposed to
have registered with it, and the server will either pass the "INVITE"
to the called device or, more infrequently, reply to you asking to send
it directly to the called device at the IP address so-and-so, much like
a web server when it sends an "HTTP redirect" (that would be the
so-called stateless proxying).
> 10000 is the sipgate test number that i am dialling and 217.10.79.219
> is the sipgate ip address and obviously 5060 is the port number.
> Obviously if i look into the messaging there is more stuff, but i cant
> work out whats going wrong.
If the INVITE is repeated four times, probably your phone doesn't
receive the "OK" from the server. This may be due to a number of
reasons: the outgoing packet can't go through the router; the
credentials (userID and password) that your phone uses to authenticate
itself to the server are wrong; or the replies from the SIP server
can't get in through the router and then arrive to the phone.
Even after this problem is solved, you might have a connecton but no
audio in one or both directions: this is usually due to problems with
the RTP packets that carry the voice data. Again, that could be due to
the router blocking them, or to the UDP port mapping done in a way
different from what phone and server have negotiated (that also depends
on whether or not phone and server abide by the rules described by
http://www.ietf.org/rfc/rfc3581.txt ... If your phone supports
"symmetric RTP", do enable it: it may enhance the chances of getting
the audio working.
If by now you are pulling your hair, take comfort in knowing that you
are not, by any means, the only one... See e.g. the debate at
http://www.isen.com/blog/2004/05/sip-was-good-idea-once.html .
> Im looking at captures with ethereal, but while i sort of understand
> what i am looking at, i dont really know what to look for with regards
> to what the phone is trying to do.
Unfortunatley Ethereal can only see the packets on the LAN side of the
NAT, but can't tell you e.g. the UDP port numbers (both source and
destination) on the external side.
> Any more help greatly appreciated!!!
This introduction, especially the sections 1.4 and 1.5, should give you
an idea of how SIP works (or is supposed to work ;-) ):
http://www.iptel.org/ser/doc/sip_intro/sip_introduction.html
This will help you making some sense of the packets sniffed by
Ethereal.
By the way, even before placing calls you should see the "REGISTER"
transactions that your phone initiates in order to let the provider's
server know its IP address and the fact that it's online. Until the
REGISTER succeeds, there is little hope that other types of
transactions (such as the INVITEs) may have better luck...
Cheers --
Enzo