[asterisk-users] getting invites to rtp ports ??
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 188.8.131.52:52734:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> From: <sip:37120116780191250 at 184.108.40.206>;tag=1872048972
> To: <sip:3712011972592181418 at 220.127.116.11>;tag=as3a52e748
> Call-ID: 1504207870-295758084-609228182
> CSeq: 1 INVITE
> WARNING: chan_sip.c:4127 retrans_pkt: Timeout on
> 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.
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