[cisco-voip] AS5300 Media address

[Matt Slaga (US)]

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><mailto: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<mailto:sip:8411647 at x.y.z.a> SIP/2.0
Call-ID: 6a34022c7c7a0d2c6234438c553eceb2 at 172.16.1.179<mailto:6a34022c7c7a0d2c6234438c553eceb2 at 172.16.1.179>
CSeq: 2 INVITE
From: <sip:test01 at x.y.z.b><mailto:sip:test01 at x.y.z.b>;tag=e002bb19cef9ffdee832
To: <sip:8411647 at x.y.z.a><mailto: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><mailto: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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