[Users] nathelper & fax = bug ?

Daniel-Constantin Mierla daniel at voice-system.ro
Thu Aug 10 13:53:31 CEST 2006


Did you used that config, or you imported parts in your config? If you 
can provide a network trace of the call, maybe we will be able to detect 
the error.

Cheers,
Daniel


On 08/10/06 14:39, Pavel D. Kuzin wrote:
> tryed with this config.
> Reinvite not handled properly.
> Can anybody provide example configs?
>
> -- 
> Pavel D.Kuzin
> System Administrator
> Nodex  ISP
> St. Petersburg, Russia
> pk at nodex.ru
> http://nodex.ru
> ----- Original Message ----- From: "Daniel-Constantin Mierla" 
> <daniel at voice-system.ro>
> To: "Hakan YASTI" <hakanyasti at gmail.com>
> Cc: <users at openser.org>
> Sent: Thursday, August 10, 2006 12:35 AM
> Subject: Re: [Users] nathelper & fax = bug ?
>
>
>> Hello,
>>
>> start with:
>>
>> http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
>>
>> The re-INVITEs should be handled there.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 08/08/06 09:38, Hakan YASTI wrote:
>>> 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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at openser.org
>>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
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>>
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