[Asterisk-Users] SIP behind NAT (*sigh*)
Florian Overkamp
florian at obsimref.com
Thu May 15 13:18:40 MST 2003
Hi guys,
sorry to be iterating this on the list once more, but I'm not able to get
this stuff to work as I'd expect. So far, I've always managed to keep it
out of NAT environments :->
My home LAN is NATed by a simple Draytek router.
In the home LAN is an ATA186 with SIP. On the internet (public) is an
Asterisk server.
I have nat=yes in the sip.conf and the connectmode is set to look for the
Via header.
Registration works like a charm, and if I dial in from the PSTN to the ATA
the phone rings properly. However, it doesn't seem to be able to start an
RTP stream or something, because once I try to dial, it gives me a
busy/congestion tone after a couple of tries (looking at the debug info).
The context is set properly, and I have tried to enable port-forwarding of
the RTP port toward the ATA, but no luck so far..
Maybe one of you has an idea ?
vectra*CLI>
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Using latest request as basis request
Sending to 130.89.224.240 : 5060 (non-NAT)
Capabilities: us - 14, them - 13, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="1e537e10"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Ignoring this request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="26d61be9"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Ignoring this request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5a6ef345"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Ignoring this request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="02af8ccd"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Ignoring this request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="02cbd912"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
Sip read: >
INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
Contact: <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=ata1-1 33968 33968 IN IP4 130.89.224.240
s=ATA186 Call
c=IN IP4 130.89.224.0
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 11 lines
Ignoring this request
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="1be02d93"
Content-Length: 0
to 130.89.224.240:5060
Sip read: >
ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
From: sip:ata1-1 at 217.114.97.249;tag=2733832243
To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
Call-ID: 855024110 at 130.89.224.240
CSeq: 1 ACK
User-Agent: Cisco ATA v2.15 ata18x (020927a)
Content-Length: 0
8 headers, 0 lines
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