[Asterisk-Users] Asterisk to sip client behindFirewall/NAT-cancall but cannot receive calls ?

[Shoval Tomer]

When I made a call using an older version I saw, using checkpoint's user
monitor that the call was indeed using RTP (somewhere between 10000 and
20000, dynamically set for each call).

After I upgraded the firmware, the entire conversation stays on the sip
port.

> -----Original Message-----
> From: Jon Lawrence [mailto:jon at lawrence.org.uk]
> Sent: Tuesday, December 14, 2004 10:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk to sip client
behindFirewall/NAT-
> cancall but cannot receive calls ?
> 
> On Tuesday 14 December 2004 15:19, Shoval Tomer wrote:
> > As far as I can remember I only opened sip and tftp ports for the
phone.
> >
> > For some reason (didn't look into it too much) the call stays with
sip
> > and doesn't use RTP.
> >
> 
> SIP is what sets up the session (ie it does session handling)
> RTP is the transport protocol that the audio uses.
> 
> If you're using SIP then you're using RTP eos.
> 
> Jon
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