tryed with this config. Reinvite not handled properly. Can anybody provide example configs?
-- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk@nodex.ru http://nodex.ru ----- Original Message ----- From: "Daniel-Constantin Mierla" daniel@voice-system.ro To: "Hakan YASTI" hakanyasti@gmail.com Cc: users@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@voice-system.ro To: "Dmitry Lyubimkov" loft@onego.ru Cc: users@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@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@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@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@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@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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