thr3ads.net - asterisk users - [Asterisk-Users] Asterisk media problem behind NAT [Aug 2004]

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Partha Sarathi
2004-Aug-26 04:25 UTC
[Asterisk-Users] Asterisk media problem behind NAT

Hello All,

  I have a media problem while using sip communicator
user agent with asterisk behind NAT.I had enabled the
debug mode in asterisk and capture the results.I have
attached the results with this mail.Can any one help
me to fix  the problem?

Thanks in advance,
Partha  


		
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-------------- next part --------------
Sip read:
REGISTER sip:<asterisk ip>:5060;transport=udp SIP/2.0
Call-ID: 18883d642b2e1730629c792a35bb0fb6@172.16.1.54
CSeq: 1 REGISTER
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: "3002" <sip:3002@<asterisk ip>:5060;transport=udp>
Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa
Max-Forwards: 70
Expires: 3600
Contact: "3002" <sip:<gateway1>:5060;transport=udp>
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 172.16.1.54 : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa;received=<gateway1>
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=as4a5aa3e3
Call-ID: 18883d642b2e1730629c792a35bb0fb6@172.16.1.54
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3002@<asterisk ip>>
Content-Length: 0


 to <gateway1>:5060
    -- Registered SIP '3002' at <gateway1> port 5060 expires 3600
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.1.54:5060;branch=z9hG4bK850e4d5c6ff86d8844678ba62b0e89aa;received=<gateway1>
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=as4a5aa3e3
Call-ID: 18883d642b2e1730629c792a35bb0fb6@172.16.1.54
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:3002@<asterisk ip>>;expires=3600
Date: Thu, 26 Aug 2004 10:33:32 GMT
Content-Length: 0


 to <gateway1>:5060


Sip read:
REGISTER sip:<asterisk ip>:5060;transport=udp SIP/2.0
Call-ID: ec142a25ca3fef012098b2782a8c7437@192.168.1.38
CSeq: 1 REGISTER
From: "3004" <sip:3004@<asterisk
ip>:5060;transport=udp>;tag=13645178
To: "3004" <sip:3004@<asterisk ip>:5060;transport=udp>
Via: SIP/2.0/UDP
192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021
Max-Forwards: 70
Expires: 3600
Contact: "3004" <sip:192.168.1.38:5060;transport=udp>
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.38 : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021;received=<gateway2
ip>
From: "3004" <sip:3004@<asterisk
ip>:5060;transport=udp>;tag=13645178
To: "3004" <sip:3004@<asterisk
ip>:5060;transport=udp>;tag=as0934b948
Call-ID: ec142a25ca3fef012098b2782a8c7437@192.168.1.38
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3004@<asterisk ip>>
Content-Length: 0


 to <gateway2 ip>:5060
    -- Registered SIP '3004' at <gateway2 ip> port 5060 expires
3600
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.38:5060;branch=z9hG4bK7ca01be9be9a2dca34277ce0ef3f5021;received=<gateway2
ip>
From: "3004" <sip:3004@<asterisk
ip>:5060;transport=udp>;tag=13645178
To: "3004" <sip:3004@<asterisk
ip>:5060;transport=udp>;tag=as0934b948
Call-ID: ec142a25ca3fef012098b2782a8c7437@192.168.1.38
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:3004@<asterisk ip>>;expires=3600
Date: Thu, 26 Aug 2004 10:33:45 GMT
Content-Length: 0


 to <gateway2 ip>:5060


Sip read:
INVITE sip:3004@<asterisk ip> SIP/2.0
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 1 INVITE
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: <sip:3004@<asterisk ip>>
Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521
Max-Forwards: 70
Contact: "3002" <sip:<gateway1>:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 148

v=0
o=par 0 0 IN IP4 <gateway1>
s=-
c=IN IP4 <gateway1>
t=0 0
m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18
m=video 22222 RTP/AVP 26 34 31

10 headers, 7 lines
Using latest request as basis request
Sending to 172.16.1.54 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found video format UNKN
Found video format UNKN
Found video format UNKN
Capabilities: us - 786446, them - 303/851968, combined - 786446
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 3004 in default
list_route: hop: <sip:<gateway1>:5060;transport=udp>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521;received=<gateway1>
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: <sip:3004@<asterisk ip>>;tag=as5bca4b71
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3004@<asterisk ip>>
Content-Length: 0


 to <gateway1>:5060
    -- Executing Dial("SIP/3002-40e9", "SIP/3004|15") in new
stack
We're at <asterisk ip> port 16112
Video is at <asterisk ip> port 18274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
Answering with non-codec capability 1
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:<gateway2 ip> SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 26 Aug 2004 10:33:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 1049 1049 IN IP4 <asterisk ip>
s=session
c=IN IP4 <asterisk ip>
t=0 0
m=audio 16112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 18274 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway2 ip>:5060
    -- Called 3004


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 INVITE
Content-Length: 0


7 headers, 0 lines


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: "3004" <sip:192.168.1.38:5060;transport=udp>
Content-Length: 144

v=0
o=vaa 0 0 IN IP4 192.168.1.38
s=-
c=IN IP4 192.168.1.38
t=0 0
m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18
m=video 22222 RTP/AVP 26 34 31

9 headers, 7 lines
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found video format UNKN
Found video format UNKN
Found video format UNKN
Capabilities: us - 786436, them - 303/851968, combined - 786436
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:192.168.1.38:5060;transport=udp>
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
Transmitting:
ACK sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to <gateway2 ip>:5060
    -- SIP/3004-b85f answered SIP/3002-40e9
We're at <asterisk ip> port 15204
Video is at <asterisk ip> port 11274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521;received=<gateway1>
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: <sip:3004@<asterisk ip>>;tag=as5bca4b71
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3004@<asterisk ip>>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1049 IN IP4 <asterisk ip>
s=session
c=IN IP4 <asterisk ip>
t=0 0
m=audio 15204 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 11274 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000

 to <gateway1>:5060
    -- Attempting native bridge of SIP/3002-40e9 and SIP/3004-b85f


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: "3004" <sip:192.168.1.38:5060;transport=udp>
Content-Length: 144

v=0
o=vaa 0 0 IN IP4 192.168.1.38
s=-
c=IN IP4 192.168.1.38
t=0 0
m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18
m=video 22222 RTP/AVP 26 34 31

9 headers, 7 lines
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found video format UNKN
Found video format UNKN
Found video format UNKN
Capabilities: us - 786436, them - 303/851968, combined - 786436
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:192.168.1.38:5060;transport=udp>
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
Transmitting:
ACK sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to <gateway2 ip>:5060


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: "3004" <sip:192.168.1.38:5060;transport=udp>
Content-Length: 144

v=0
o=vaa 0 0 IN IP4 192.168.1.38
s=-
c=IN IP4 192.168.1.38
t=0 0
m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18
m=video 22222 RTP/AVP 26 34 31

9 headers, 7 lines
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found video format UNKN
Found video format UNKN
Found video format UNKN
Capabilities: us - 786436, them - 303/851968, combined - 786436
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:192.168.1.38:5060;transport=udp>
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
Transmitting:
ACK sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to <gateway2 ip>:5060


Sip read:
ACK sip:3004@<asterisk ip>;transport=udp SIP/2.0
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 1 ACK
From: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
To: <sip:3004@<asterisk ip>>;tag=as5bca4b71
Via: SIP/2.0/UDP 172.16.1.54:5060;branch=3137322e31362e312e35343a3530363
Max-Forwards: 70
Content-Length: 0


8 headers, 0 lines
set_destination: Parsing <sip:<gateway1>:5060;transport=udp> for
address/port to send to
set_destination: set destination to <gateway1>, port 5060
We're at <asterisk ip> port 15204
Video is at <asterisk ip> port 11274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:<gateway1>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK488229de
From: <sip:3004@<asterisk ip>>;tag=as5bca4b71
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
Contact: <sip:3004@<asterisk ip>>
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1049 1050 IN IP4 192.168.1.38
s=session
c=IN IP4 192.168.1.38
t=0 0
m=audio 22224 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway1>:5060
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
We're at <asterisk ip> port 16112
Video is at <asterisk ip> port 18274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6c
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1050 IN IP4 <gateway1>
s=session
c=IN IP4 <gateway1>
t=0 0
m=audio 22224 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway2 ip>:5060
set_destination: Parsing <sip:<gateway1>:5060;transport=udp> for
address/port to send to
set_destination: set destination to <gateway1>, port 5060
We're at <asterisk ip> port 15204
Video is at <asterisk ip> port 11274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:<gateway1>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK488229df
From: <sip:3004@<asterisk ip>>;tag=as5bca4b71
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
Contact: <sip:3004@<asterisk ip>>
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1049 1051 IN IP4 192.168.1.38
s=session
c=IN IP4 192.168.1.38
t=0 0
m=audio 22224 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway1>:5060
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
We're at <asterisk ip> port 16112
Video is at <asterisk ip> port 18274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6d
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1051 IN IP4 <gateway1>
s=session
c=IN IP4 <gateway1>
t=0 0
m=audio 22224 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway2 ip>:5060


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6c
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 103 INVITE
Content-Length: 0


7 headers, 0 lines


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6d
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 104 INVITE
Content-Length: 0


7 headers, 0 lines


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK488229de
From: <sip:3004@<asterisk ip>>;tag=as5bca4b71
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 102 INVITE
Content-Length: 0


7 headers, 0 lines


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK488229df
From: <sip:3004@<asterisk ip>>;tag=as5bca4b71
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 103 INVITE
Content-Length: 0


7 headers, 0 lines
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
We're at <asterisk ip> port 16112
Video is at <asterisk ip> port 18274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6e
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1052 IN IP4 <gateway1>
s=session
c=IN IP4 <gateway1>
t=0 0
m=audio 15204 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway2 ip>:5060


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6e
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 105 INVITE
Content-Length: 0


7 headers, 0 lines
set_destination: Parsing <sip:<gateway1>:5060;transport=udp> for
address/port to send to
set_destination: set destination to <gateway1>, port 5060
We're at <asterisk ip> port 15204
Video is at <asterisk ip> port 11274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:<gateway1>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK488229e0
From: <sip:3004@<asterisk ip>>;tag=as5bca4b71
To: "3002" <sip:3002@<asterisk
ip>:5060;transport=udp>;tag=24957277
Contact: <sip:3004@<asterisk ip>>
Call-ID: 24d4b87f5d2fef99011670bdf2152c66@172.16.1.54
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1052 IN IP4 <gateway2 ip>
s=session
c=IN IP4 <gateway2 ip>
t=0 0
m=audio 16112 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 22222 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway1>:5060
set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.38, port 5060
We're at <asterisk ip> port 16112
Video is at <asterisk ip> port 18274
Answering with preferred capability 4
Answering with preferred capability 262144
Answering with preferred capability 524288
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6f
From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5
To: <sip:<gateway2 ip>>;tag=33114244
Contact: <sip:3002@<asterisk ip>>
Call-ID: 7e8dece54154cc8428630434213e4369@<asterisk ip>
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 1049 1053 IN IP4 <gateway1>
s=session
c=IN IP4 <gateway1>
t=0 0
m=audio 15204 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 11274 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
 (NAT) to <gateway2 ip>:5060
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asterisk users - Aug 2004 - Asterisk media problem behind NAT

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