[Users] nathelper & fax = bug ?

Pavel D. Kuzin pk at nodex.ru
Thu Aug 10 13:39:06 CEST 2006


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