[cisco-voip] Re: ATA : NAT problem

[Oleg Shevtsov]

You should fill your NATSERVER and NATTIMER to allow ATA sending
KeepAlive messages. Better to use your SIP server as NATSERVER.


On Wed, Sep 15, 2004 at 10:14:10AM +0200, Nicolas RUIZ wrote:
>Hi,
>
>I have a Cisco ATA 186 (code 3.1.0) with NAT, I have set :
>
>NATIP: 62.39.70.245
>NATSERVER : 0
>NATTIMER: 0x00000000
>
>When, I sniff the network where is my ATA, I don't receive any SIP frames.
>
>I have probably a NAT problem;
>
>THIS IS A DEBUG ON MY SIP SERVER :
>
>IP at WAN : 62.39.70.245
>IP@ SIP SERVER : 80.118.128.5
>
>U 80.118.128.5:5060 -> 62.39.70.245:5060
>  INVITE sip:0170708661 at 62.39.70.245:5060;user=phone;transport=udp
>SIP/2.0..Record-Route: <sip:0170708661 at 80.118.128.5;ftag=22002
>  DE0-3F0;lr>..Via: SIP/2.0/UDP
>80.118.128.5;branch=z9hG4bK67ae.d6569395.0..Via: SIP/2.0/UDP
>80.118.128.1:5060..From: <sip:01707
>  08661 at 80.118.128.1>;tag=22002DE0-3F0..To:
><sip:0170708661 at sip.vivaction.net>..Date: Tue, 14 Sep 2004 11:45:10
>GMT..Call-ID: 5E0
>  BE2D0-57A11D9-8505945B-39C57B0D at 80.118.128.1..Supported:
>timer,100rel..Min-SE:  1800..Cisco-Guid: 1577637232-91886041-223153877
>  9-969243405..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE,
>OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, N
>  OTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID:
><sip:0170708661 at 80.118.128.1>;party=calling;screen=yes;privacy
>  =off..Timestamp: 1095162310..Contact:
><sip:0170708661 at 80.118.128.1:5060>..Expires: 180..Allow-Events:
>telephone-event..Content-
>  Type: application/sdp..Content-Length:
>291....v=0..o=CiscoSystemsSIP-GW-UserAgent 5743 1810 IN IP4
>80.118.128.1..s=SIP Call..c=
>  IN IP4 80.118.128.1..t=0 0..m=audio 17738 RTP/AVP 4 101 19..c=IN IP4
>80.118.128.1..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=no..a=
>  rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:19
>CN/8000..a=ptime:30..
>########################
>U 80.118.128.5:5060 -> 62.39.70.245:5060
>  INVITE sip:0170708661 at 62.39.70.245:5060;user=phone;transport=udp
>SIP/2.0..Record-Route: <sip:0170708661 at 80.118.128.5;ftag=22002
>  DE0-3F0;lr>..Via: SIP/2.0/UDP
>80.118.128.5;branch=z9hG4bK67ae.d6569395.0..Via: SIP/2.0/UDP
>80.118.128.1:5060..From: <sip:01707
>  08661 at 80.118.128.1>;tag=22002DE0-3F0..To:
><sip:0170708661 at sip.vivaction.net>..Date: Tue, 14 Sep 2004 11:45:10
>GMT..Call-ID: 5E0
>  BE2D0-57A11D9-8505945B-39C57B0D at 80.118.128.1..Supported:
>timer,100rel..Min-SE:  1800..Cisco-Guid: 1577637232-91886041-223153877
>  9-969243405..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE,
>OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, N
>  OTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID:
><sip:0170708661 at 80.118.128.1>;party=calling;screen=yes;privacy
>  =off..Timestamp: 1095162310..Contact:
><sip:0170708661 at 80.118.128.1:5060>..Expires: 180..Allow-Events:
>telephone-event..Content-
>  Type: application/sdp..Content-Length:
>291....v=0..o=CiscoSystemsSIP-GW-UserAgent 5743 1810 IN IP4
>80.118.128.1..s=SIP Call..c=
>  IN IP4 80.118.128.1..t=0 0..m=audio 17738 RTP/AVP 4 101 19..c=IN IP4
>80.118.128.1..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=no..a=
>  rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:19
>CN/8000..a=ptime:30..
>
>But when i set NATIP: 0.0.0.0
>
>I can do make outgoing calls but, i cannnot receive call because, the proxy
>sip send to the IP@ private of the ATA.
>
>Thanks a lot
>
>Nicolas RUIZ
>os Solutions Voix-Data !
>
>STC
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-- 
Oleg Shevtsov
Chief of NOC InterExchange Carrier
phone +380442399740
mobile +380672341642
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