Connection scheme:
UA - router with NAT - OpenSER with nathelper - PSTN
gateway (Cisco AS5350)
(192.168.13.109) (217.107.59.194) (62.33.22.14)
(62.33.22.11)
Both incoming and outgoing calls work right. Openser uses the
nathelper
module for proxing of rtp stream of NAT UA.
Here is example of SIP messages (call from PSTN through a gateway):
15:37:07.406529 IP 62.33.22.11.54581 > 62.33.22.14.5060: UDP, length
1121
E..}........>!..>!...5...i.hINVITE sip:[EMAIL PROTECTED]:5060
SIP/2.0
Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
From: <sip:[EMAIL PROTECTED]>;tag=A515D068-227D
To: <sip:[EMAIL PROTECTED]>
Date: Fri, 04 Aug 2006 11:37:07 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1688609156-585372123-2332753710-1711444852
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID:
<sip:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1154691427
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 316
v=0
o=CiscoSystemsSIP-GW-UserAgent 4330 9654 IN IP4 62.33.22.11
s=SIP Call
c=IN IP4 62.33.22.11
t=0 0
m=audio 17088 RTP/AVP 3 18 8 0 4
c=IN IP4 62.33.22.11
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
Nathelper works right and in the message sent to UA you can see
already
IP address of Openser (62.33.22.14) instead of the address of a
gateway
(62.33.22.11):
15:37:07.407463 IP 62.33.22.14.5060 > 217.107.59.194.47331: UDP,
length
1256
[EMAIL PROTECTED]@..|>!...k;.......n^INVITE sip:[EMAIL PROTECTED]:47331
SIP/2.0
Record-Route: <sip:62.33.22.14;lr;ftag=A515D068-227D>
Via: SIP/2.0/UDP voapp.ru:5060;branch=z9hG4bK2d06.d63c8585.0
Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
From: <sip:[EMAIL PROTECTED]>;tag=A515D068-227D
To: <sip:[EMAIL PROTECTED]>
Date: Fri, 04 Aug 2006 11:37:07 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1688609156-585372123-2332753710-1711444852
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID:
<sip:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1154691427
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 334
v=0
o=CiscoSystemsSIP-GW-UserAgent 4330 9654 IN IP4 62.33.22.11
s=SIP Call
c=IN IP4 62.33.22.14
t=0 0
m=audio 35858 RTP/AVP 3 18 8 0 4
c=IN IP4 62.33.22.14
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=nortpproxy:yes
After some talking the subscriber from PSTN tries to send a fax.
PSTN gateway detects it and sends this message:
15:37:22.512722 IP 62.33.22.11.51655 > 62.33.22.14.5060: UDP, length
1276
E..........z>!..>!..........INVITE
sip:62.33.22.14:5060;from-tag=A515D068-227D;lr SIP/2.0
Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
From: <sip:[EMAIL PROTECTED]>;tag=A515D068-227D
To: <sip:[EMAIL PROTECTED]>;tag=bbaac0e818284ff5
Date: Fri, 04 Aug 2006 11:37:22 GMT
Call-ID: [EMAIL PROTECTED]
Route: <sip:[EMAIL PROTECTED]:47331>
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1688609156-585372123-2332753710-1711444852
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 102 INVITE
Max-Forwards: 6
Remote-Party-ID:
<sip:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1154691442
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 393
v=0
o=CiscoSystemsSIP-GW-UserAgent 4330 9656 IN IP4 62.33.22.11
s=SIP Call
c=IN IP4 62.33.22.11
t=0 0
m=image 17088 udptl t38
c=IN IP4 62.33.22.11
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
Openser processes is and sends to UA:
15:37:22.513017 IP 62.33.22.14.5060 > 217.107.59.194.47331: UDP,
length
1336
[EMAIL PROTECTED]@..,>!...k;[EMAIL PROTECTED] sip:[EMAIL PROTECTED]:47331
SIP/2.0
Record-Route: <sip:62.33.22.14;lr;ftag=A515D068-227D>
Via: SIP/2.0/UDP voapp.ru:5060;branch=z9hG4bKfc06.4b118272.0
Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
From: <sip:[EMAIL PROTECTED]>;tag=A515D068-227D
To: <sip:[EMAIL PROTECTED]>;tag=bbaac0e818284ff5
Date: Fri, 04 Aug 2006 11:37:22 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1688609156-585372123-2332753710-1711444852
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
CSeq: 102 INVITE
Max-Forwards: 5
Remote-Party-ID:
<sip:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1154691442
Contact: <sip:[EMAIL PROTECTED]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 393
v=0
o=CiscoSystemsSIP-GW-UserAgent 4330 9656 IN IP4 62.33.22.11
s=SIP Call
c=IN IP4 62.33.22.11
t=0 0
m=image 17088 udptl t38
c=IN IP4 62.33.22.11
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
As you can see the nathelper module has not worked since the field
c=IN
IP4 62.33.22.11 has not changed.
Probably it has taken place because m=image instead of m=audio as
usual.
As a result of transfer of a fax has not taken place.
If to place UA outside for NAT router all works that once again
confirms
that bug is in the nathelper module.
Questions:
Why the module behaves so? What difference that to proxing (what
byte stream and in what format)?
How it can be bypassed?
Also that the most interesting - UA refuses to accept T38 and
suggests
to use instead of it G.711 codec and the gateway agrees i.e. in
result
we have audio stream.
Dmitry
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