I’m using Using Trixbox.
I connect to a asterisk server from a remote client that runs behind a firewall. This client fowards the sip and rtp ports, which in this case are 5070 and rtp starting at 10000. The problem I believe I’m running into is that no matter what I do with the sip.conf entry for this peer, it still resolves the external ip address to the internal ip address of the remote machine in the remote network, noted in the sip debug output…
e.g. Peer audio RTP is at port 192.168.0.4:10000 ( This is an internal ip )
I was under the impression that even if the SIP packet contains the internal ip address of the remote network, it the nat setting was set to nat=yes it would use the ipaddress of where the packet is coming from, however this does not seem to be the case for this configuration.
I also have a xlite softphone on the remote machine and it works as expected and reports:
Peer audio RTP is at port 65.6.204.130:10027
I’m not sure how to proceed as I’m not yet completely familiar with the sip debug and the information it’s conveying. Any help would be appreciated.
*********************** SIP.CONF **********************
username=4004
type=friend
secret=*****
record_out=Always
record_in=Always
qualify=yes
nat=yes
mailbox=4004@device
host=65.6.204.130
ipaddr=65.6.204.130
port=5070
dtmfmode=rfc2833
context=from-internal
allow=all
insecure=very
canreinvite=no
caninvite=yes
callerid=“Incoming Skype”<305-388-1876>
*********************** SIP DEBUG**********************
<-- SIP read from 65.6.204.130:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK23ea724c;rport
To: sip:+17864234600@65.6.204.130:5070;tag=4679
From: “Unknown” sip:Unknown@127.0.0.1;tag=as68ee9aeb
Call-ID: 0f1abc11096650f60fa8c8f776023722@127.0.0.1
CSeq: 103 INVITE
User-Agent: Uplink Skype2Sip 1.21
Contact: sip:4004@65.6.204.130:5070
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 342
v=0
o=- 823474570 823474596 IN IP4 65.6.204.130
s=Uplink Skype2Sip
c=IN IP4 192.168.0.4
t=0 0
m=audio 10000 RTP/AVP 0 8 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.0.4 10000
a=domain:65.6.204.130
— (13 headers 16 lines)—
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.4:10000
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x1e (gsm|ulaw|alaw|g726)/video=0x0 (nothing), combined - 0x1e (gsm|ulaw|alaw|g726)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: sip:4004@65.6.204.130:5070
set_destination: Parsing sip:4004@65.6.204.130:5070 for address/port to send to
set_destination: set destination to 65.6.204.130, port 5070
Transmitting (NAT) to 65.6.204.130:5070:
ACK sip:4004@65.6.204.130:5070 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK10fc599e;rport
From: “Unknown” sip:Unknown@127.0.0.1;tag=as68ee9aeb
To: sip:+17864234600@65.6.204.130:5070;tag=4679
Contact: sip:Unknown@127.0.0.1
Call-ID: 0f1abc11096650f60fa8c8f776023722@127.0.0.1
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0