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
On Fri, 3 Jun 2005, 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.Turn off reinvite or whatever it's called in sip (can never remember if reinvite is the sip or iax2 term). This will force the audio through asterisk. -forrest
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 > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
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 <asterisk-users@citrini.com> Subject: Re: [Asterisk-Users] Sip UA behind NAT To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <42A0657E.3080604@citrini.com> 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 >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >