[Users] nathelper & fax = bug ?

Pavel D. Kuzin pk at nodex.ru
Thu Aug 10 13:53:48 CEST 2006


i`m tryed to user exacly equals config.
http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
and 2 users refistered with openser.

--
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: "Pavel D. Kuzin" <pk at nodex.ru>
Cc: "Hakan YASTI" <hakanyasti at gmail.com>; <users at openser.org>
Sent: Thursday, August 10, 2006 3:53 PM
Subject: Re: [Users] nathelper & fax = bug ?


> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at openser.org
>>>>> http://openser.org/cgi-bin/mailman/listinfo/users 
>>>>
>>>>
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>>>>
>>>
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