[cisco-voip] AS5300 Media address

[Peter Slow]

Alternatively, put a CUBE at your NAT/PAT boundary, either AS the
boundary or in paralell to it.

On Tue, Apr 6, 2010 at 2:04 PM, Matt Slaga (US)
<Matt.Slaga at us.didata.com> wrote:
> Some other firewalls call it ‘ALG’, application layer gateway.
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
> Sent: Tuesday, April 06, 2010 1:55 PM
> To: David Eco
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] AS5300 Media address
>
>
>
> I believe the NAT device in your scenario is not performing fixup.  NAT
> involves rewriting not only the IP header but also fixing up the message
> body to rewrite embedded IPs and ports.  This may involve dynamic port
> allocation as well.  It is generally referred to as 'protocol fixup' on the
> NAT device.
>
> /Wes
>
> On Tuesday, April 06, 2010 12:15:15 PM, David Eco <david.eco at msn.com> wrote:
>
> Hello,
> I tried to send the SIP call to AS5350 from a IP PBX but got one way audio.
> The scenario is SIP phone ->NAT->IP PBX -> AS5350 ->T1 circuit.
> When the call reached AS5350, it came with private IP address and sent the
> RTP back to it instead of public IP's.
> Is there any way to set the media address as the IP's in FROM which means
> sending the media back to the origination, IP PBX? Thank you.
>
>
>
> ======Debug output=======
> (x.y.z.a=AS5350   x.y.z.b=IP PBX)
>
> Apr  6 14:15:23.866: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:8411647 at x.y.z.a SIP/2.0
> Call-ID: 6a34022c7c7a0d2c6234438c553eceb2 at 172.16.1.179
> CSeq: 2 INVITE
> From: <sip:test01 at x.y.z.b>;tag=e002bb19cef9ffdee832
> To: <sip:8411647 at x.y.z.a>
> Via: SIP/2.0/UDP x.y.z.b:5061;branch=z9hG4bk-e002bb19cef9ffdee832
> Max-Forwards: 70
> Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
> Contact: <sip:ippbx at x.y.z.b:5061>
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 339
>
> v=0
> o=CMI-SIPUA 63439 0 IN IP4 172.16.1.179
> s=SIP CALL
> c=IN IP4 172.16.1.179
> t=0 0
> m=audio 60000 RTP/AVP 0 8 4 18 23 22 2 21 101
> a=rtpmap:23 G726-16/8000
> a=rtpmap:22 G726-24/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:21 G726-40/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=fmtp:18 annexb=no
> a=rtcp:60001
> a=sendrecv
>
>
>
>           Stream type            : voice-only
>           Media line             : 1
>           State                  : STREAM_ADDING (2)
>           Callid                 : -1
>           Negotiated Codec       : g711ulaw, bytes :160
>           Negotiated DTMF relay  : inband-voice
>           Negotiated NTE payload : 0
>           Negotiated CN payload  : 0
>           Media Srce Addr/Port   : x.y.z.a
>           Media Dest Addr/Port   : 172.16.1.179:44042
>
>
>
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