[asterisk-users] SIP RTP Tunnel

[kalle.odenthal at genion.de]

Thank you Salvatore,
I have Wireshark ;) But my problem is not to SEE the packets.
I'm trying to do a Handover between VoIP and GSM using an Asterisk Server. But to avoid the effort to implement the channel: Asterisk->public switched network->GSM->...radio...->GSM modem-> SmartPhone, I want to substitute the "GSM" Channel with SIP. This is valid for my work. The problem is if Asterisk DO tunnel all Packets through its bones when I switch between SIP and GSM, it connects two SIP participants direcly. 
You see my problem?

Saludos, 

Kalle 


You should get a packet capture and look at the SDP that is agreed to by both parties. It sounds like someone is not honoring it.

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com
 
VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
     Fax: (212) 279-2906
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of kalle.odenthal at genion.de
Sent: Thursday, March 29, 2007 9:35 PM
To: asterisk-users at lists.digium.com
Subject: RE: RE: [asterisk-users] SIP RTP Tunnel

Hola Sanjay, 

this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1.

Any idea?

Thanx!!

-----Original Message-----
From: Sanjay Rajdev [mailto:sanjay.rajdev at featherstoneinformatics.com]
Sent: Donnerstag, 29. März 2007 18:27
To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] SIP RTP Tunnel

Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting).

Regards,
Sanjay Rajdev

----- Original Message -----
From: "kalle odenthal" <kalle.odenthal at genion.de>
To: asterisk-users at lists.digium.com
Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] SIP RTP Tunnel

Hello,

is it possible to rout ALL RTP Data over Asterisk, like

SIP1 <---RTP---> Asterisk <---RTP---> SIP2

I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) 

Thanx, 

Kalle




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