[Users] nathelper & fax = bug ?

Tue Aug 8 08:38:22 CEST 2006

Hi,
Is there anybody who will share his config file,( or a samle configuration ) 
which is working properly with rtp_proxy or mediaproxy ? ( handle re-INVITEs 
properly ).
As I see, there are some people have the same problem,like me.
Thanks,

----- Original Message ----- 
From: "Daniel-Constantin Mierla" <daniel at voice-system.ro>
To: "Dmitry Lyubimkov" <loft at onego.ru>
Cc: <users at openser.org>
Sent: Monday, August 07, 2006 11:12 PM
Subject: Re: [Users] nathelper & fax = bug ?

> Hello,
>
> the latest openser should not care about type of media (audio or image is 
> same for openser). The problem is that you do not force the rtpproxy for 
> re-INVITE in your config file, but only for initial INVITE of the call.
>
> Cheers,
> Daniel
>
>
> On 08/05/06 10:52, Dmitry Lyubimkov wrote:
>> 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:78142799233 at voapp.ru:5060 SIP/2.0
>> Via: SIP/2.0/UDP  62.33.22.11:5060;x-route-tag="tgrp:ipphone"
>> From: <sip:78142764164 at 62.33.22.11>;tag=A515D068-227D
>> To: <sip:78142799233 at voapp.ru>
>> Date: Fri, 04 Aug 2006 11:37:07 GMT
>> Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374 at 195.161.136.114
>> 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:78142764164 at 62.33.22.11>;party=calling;screen=yes;privacy=off
>> Timestamp: 1154691427
>> Contact: <sip:78142764164 at 62.33.22.11:5060>
>> 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
>> E..... at .@..|>!...k;.......n^INVITE sip:ngul at 217.107.59.194:47331 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:78142764164 at 62.33.22.11>;tag=A515D068-227D
>> To: <sip:78142799233 at voapp.ru>
>> Date: Fri, 04 Aug 2006 11:37:07 GMT
>> Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374 at 195.161.136.114
>> 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:78142764164 at 62.33.22.11>;party=calling;screen=yes;privacy=off
>> Timestamp: 1154691427
>> Contact: <sip:78142764164 at 62.33.22.11:5060>
>> 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:78142764164 at 62.33.22.11>;tag=A515D068-227D
>> To: <sip:78142799233 at voapp.ru>;tag=bbaac0e818284ff5
>> Date: Fri, 04 Aug 2006 11:37:22 GMT
>> Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374 at 195.161.136.114
>> Route: <sip:ngul at 217.107.59.194:47331>
>> 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:78142764164 at 62.33.22.11>;party=calling;screen=yes;privacy=off
>> Timestamp: 1154691442
>> Contact: <sip:78142764164 at 62.33.22.11:5060>
>> 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
>> E..T.. at .@..,>!...k;...... at n.INVITE sip:ngul at 217.107.59.194:47331 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:78142764164 at 62.33.22.11>;tag=A515D068-227D
>> To: <sip:78142799233 at voapp.ru>;tag=bbaac0e818284ff5
>> Date: Fri, 04 Aug 2006 11:37:22 GMT
>> Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374 at 195.161.136.114
>> 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:78142764164 at 62.33.22.11>;party=calling;screen=yes;privacy=off
>> Timestamp: 1154691442
>> Contact: <sip:78142764164 at 62.33.22.11:5060>
>> 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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>>
>>
>
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