[asterisk-users] getting invites to rtp ports ??
Joshua Colp
jcolp at digium.com
Wed Aug 29 08:40:25 CDT 2018
On Wed, Aug 29, 2018, at 10:34 AM, sean darcy wrote:
> I'm getting invites to very high ports every 30 seconds from a
> particular ip address:
>
> Retransmitting #10 (NAT) to 5.199.133.128:52734:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
> From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972
> To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748
> Call-ID: 1504207870-295758084-609228182
> CSeq: 1 INVITE
> .......
> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
> 1504207870-295758084-609228182...
>
> I thought invites had to go to port 5060 or so. I don't understand why
> somebody (let's assume a bad guy) is trying ports above 50000.
There is nothing that explicitly states that it has to be 5060, and in the case of the above it's just a random source port.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
More information about the asterisk-users
mailing list