3 Sep
2003
3 Sep
'03
4:55 a.m.
Hi,
I have come across a similar problem. I want to use the A/V facilities of
Windows Messenger if one or both of my clients are behind a NAT. I realise
that it seems to be impossible to get this to work if the NAT is symmetric.
(See: RFC3489 "Applicability Statement")
If only one of my clients is behind a NAT then it would seem I would have to
communicate my NATs external address and port mapping to the non-NAT'd
client (possibly with the help of STUN) in my SIP Invite SDP message. I
would also have to setup UDP mappings for SIP, RTP/RTCP Audio Video on my
NAT. Are my A/V port mappings also included in the Invite SDP message?
If both clients are NAT'd then what is the approach? I don't see how I
register with the SIP server using an external NAT address (my guess is this
is what I would have to use if I wanted anyone on the other side of the NAT
to be able to see me). MSFT have seemed to got around this problem by
recommended everyone to use uPnP enabled NATs, which will automatically bind
to an external address on the NAT and, I assume, use this when they register
with the SIP server.
RTPProxy is here https://demo.portaone.com/~sobomax/PortaSIP/ how does
RTPProxy help in the NAT situation, does it at all?
Please correct me where I am wrong, I am still trying to get my head around
all of this! Has anyone successfully been able to get any of the scenarios
above working (with AV)?
Regards,
Gary Brewer
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: 03 September 2003 00:35
To: Dan Fernandez
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] No audio
Hello,
is one (or both) phones behind a NAT ? If so you will need to configure
your nat to let the media traffic through.
If not, then we will need more information.
Jan.
On 02-09 18:55, Dan Fernandez wrote:
Greetings from So. America!
I just begun playing with SER. So far I can place calls within a LAN. I
have SER running on a box, with one public and one private IP. My problem
right now is that if I try to call from a UA connected via a dialup, there
is no audio in either end. The setup of the call seems fine (the phones
ring
OK). I have disable the firewall and still no luck. Do
I need to install
rtpproxy? Where can I download it?(the links I´ve seen don´t work)
Any help would be greatly appreciated.
Rgds
Dan
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3 Sep
3 Sep
5:33 a.m.
Hello, comments inline. All the NAT stuff is quite complicated and
whether it would work or not depends on many factors.
On 03-09 09:55, Gary Brewer wrote:
Hi,
I have come across a similar problem. I want to use the A/V facilities of
Windows Messenger if one or both of my clients are behind a NAT. I realise
that it seems to be impossible to get this to work if the NAT is symmetric.
(See: RFC3489 "Applicability Statement")
That depends. It could work even with symmetric NAT if the client you
are using does support symmetric signalling and RTP (I am not sure Windows
Messenger is). That means the user agent must be ready to receive SIP
requests and responses on the same port which was used as the source port
for sending SIP messages. Also it must support symmetric RTP to make
media work.
Also the user agent must create REGISTERs containing public IP of the
NAT (can be determined using STUN, for example), or you would have to
use nathelper module on the server.
If only one of my clients is behind a NAT then it
would seem I would have to
communicate my NATs external address and port mapping to the non-NAT'd
client (possibly with the help of STUN) in my SIP Invite SDP message. I
Yes, but Windows Messenger doesn't support STUN.
would also have to setup UDP mappings for SIP,
RTP/RTCP Audio Video on my
NAT. Are my A/V port mappings also included in the Invite SDP message?
In case of symmetrict RTP, the client in the public internet will
ignore what it receives in SDP and will send media back to the same IP
and port from which it comes from the other side. That makes the
communication through a NAT possible, but the client behind the NAT
must send first media package (that packet will open a pinhole in the
NAT). Also both sides must support symmetrict RTP and the client
behind the NAT must signal that it is using this approach.
If the clients do not support symmetric signalling, then some kind of
"NAT configuration" would be necesarry.
For example, many user agents can be configured to use ports from a
specified port range only for RTP (let's say ports 10000-10100). You
can then configure your NAT box to forward all the ports from the
range back to your user agent.
If both clients are NAT'd then what is the
approach? I don't see how I
register with the SIP server using an external NAT address (my guess is this
is what I would have to use if I wanted anyone on the other side of the NAT
to be able to see me). MSFT have seemed to got around this problem by
recommended everyone to use uPnP enabled NATs, which will automatically bind
to an external address on the NAT and, I assume, use this when they register
with the SIP server.
If both clients are behind NATs then you would probably have to use an
RTP proxy which will be placed in the public internet.
RTPProxy is here
https://demo.portaone.com/~sobomax/PortaSIP/ how does
RTPProxy help in the NAT situation, does it at all?
See the previous insertion. It could help in the case when both
clients are behind NATs and it could possibly help in some other cases
too.
Please correct me where I am wrong, I am still trying
to get my head around
all of this! Has anyone successfully been able to get any of the scenarios
above working (with AV)?
There is a couple of people who are running their clients behind NATs
on the mailing list, so I hope they could give you some advice.
Jan.
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Jan Janak