Eric, the SIP/RTP protocol does not inherently work well in NAT'd
environments. There are several commerical solutions out there to help
users traverse NAT routers and firewalls successfully, with varying
levels of success. I find that many commercial nat routers for home
users (most notably Linksys) tend to lose their nat port mappings
occasionally, causing the UA to miss incoming calls. One such
commercial company you could investigate is:
http://jasomi.com/index2.html
Since you mention that the NAT'd user is using a software UA, I might
recommend you instead investigate using the IAX protocol and an IAX
software client (such as Firefly), the IAX protocol muxes both the call
setup/teardown messenging and the real time voice traffic into a single
port, which easily traverses bi-directionally through nat routers. This
assumes that the IAX UA client generates some level of traffic every few
minutes to keep the nat router port translation mapping active - this is
usally done by having the client re-register with the registration
server (asterisk in this case) every few minutes. The IAX protocol was
developed by your same friendly Asterisk developers, and is currently
being groomed for submission to the IETF for RFC.
A brief comparison between SIP and IAX can be found here:
http://www.voip-info.org/wiki-IAX+versus+SIP
A list of hardware and software IAX UA's can be found here:
http://www.voip-info.org/wiki-Asterisk+IAX+clients
-mike
Eric Yu-Wei Sung wrote:
Hi, is there any way I could make this work without having to explicitly
perform port forwarding for RTP traffic at my NAT? (i.e. NAT
transparently sets up the RTP channel for the internal SIP UA with the
external SIP UA) Thanks Eric Date: Fri, 03 Jun 2005 09:13:18 -0500 From:
Mike Holloway <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users]
Sip UA behind NAT To: Asterisk Users Mailing List - Non-Commercial
Discussion <[email protected]> Message-ID:
<[EMAIL PROTECTED]> Content-Type: text/plain;
charset=ISO-8859-1; format=flowed Eric, The problem you are seeing is
because the RTP (voice) packets being sent towards the NAT'd UA are
being blocked by the NAT router. The UA being used behind NAT will need
to have a static IP address set (e.g. 192.168.1.50) and on the NAT
router you will need to permanently forward (port forward) SIP and RTP
ports to the internal IP address. I recommend ports 5060 and
16384-16400. On the NAT'd UA, set the SIP port to 5060 and the RTP ports
to 16384-16400. If your UA only supports one RTP port, just use 16384.
As Forrest noted, you will also want to set canreinvite=no in sip.conf
for the NAT'd UA. You should also set nat=yes, which will force asterisk
to re-write SIP packets coming from the NAT'd UA to the correct external
IP address of the NAT router. -mike Eric Yu-Wei Sung wrote:
I am trying to make 1 soft SIP UA behind NAT connect to a public hard
CISCO UA via a public asterisk server. The CISCO UA can hear the
voice from the SIP UA but not vice versa. I do set nat to yes for the
soft phone. Any help would be greatly appreciated.
Below is my sip.conf
[general]
port = 8060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[2000] ; soft phone behind NAT
type=friend ; This device takes and makes calls
username=2000 ; Username on device
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
nat=yes
[2002] ; CISCO hard phone
type=friend
username=2002
secret=2002
host=dynamic
context=from-sip
mailbox=103
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