9 Aug
2006
9 Aug
'06
11:35 p.m.
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(a)voice-system.ro>
To: "Dmitry Lyubimkov" <loft(a)onego.ru>
Cc: <users(a)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:
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http://openser.org/cgi-bin/mailman/listinfo/users
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@voapp.ru:5060
SIP/2.0
Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
From: <sip:78142764164@62.33.22.11>;tag=A515D068-227D
To: <sip:78142799233@voapp.ru>
Date: Fri, 04 Aug 2006 11:37:07 GMT
Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)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@62.33.22.11>;party=calling;screen=yes;privacy=off
Timestamp: 1154691427
Contact: <sip:78142764164@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.....@.@..|>!...k;.......n^INVITE sip:ngul@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@62.33.22.11>;tag=A515D068-227D
To: <sip:78142799233@voapp.ru>
Date: Fri, 04 Aug 2006 11:37:07 GMT
Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)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@62.33.22.11>;party=calling;screen=yes;privacy=off
Timestamp: 1154691427
Contact: <sip:78142764164@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@62.33.22.11>;tag=A515D068-227D
To: <sip:78142799233@voapp.ru>;tag=bbaac0e818284ff5
Date: Fri, 04 Aug 2006 11:37:22 GMT
Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)195.161.136.114
Route: <sip:ngul@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@62.33.22.11>;party=calling;screen=yes;privacy=off
Timestamp: 1154691442
Contact: <sip:78142764164@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..@.@..,>!...k;......@n.INVITE sip:ngul@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@62.33.22.11>;tag=A515D068-227D
To: <sip:78142799233@voapp.ru>;tag=bbaac0e818284ff5
Date: Fri, 04 Aug 2006 11:37:22 GMT
Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)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@62.33.22.11>;party=calling;screen=yes;privacy=off
Timestamp: 1154691442
Contact: <sip:78142764164@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 mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users