There is such a tool. There is a lower capability free version and a paid version. VoIPMonitor <https://www.voipmonitor.org/> Tough to say what might be going on. On the face it seems like the re invite doesn't know where to go as there is no NAT table connection entry from that port to the internal IP.
On Thu, Dec 3, 2020 at 6:13 PM Chuck McCown via AF <[email protected]> wrote: > I used to think SIP was OK. I even helped GenBand (or whatever the > Northern > Telecom manufacturer was called after they were Northern Telecom ) fix > their > SIP stack. Took a ton of wire sharking and deep packet inspection but > they > had a fault. > > But I have forgotten everything I ever knew about SIP. I am sure there is > someone that makes a SIP tool the is like wire shark but decodes things > into > a much more easy to read format and then is able to launch SIP packets > too... but too lazy to see if one exists. A SIP simulator, or spoofer or > stingray type of box. > > Needs to be made out of metal and have buttons, lights and meters on it. > > -----Original Message----- > From: Ken Hohhof > Sent: Thursday, December 3, 2020 4:57 PM > To: 'AnimalFarm Microwave Users Group' > Subject: Re: [AFMUG] Offnet SIP and Multi-homing > > I thought re-invites were used to get the media stream to flow directly > between the endpoints and bypass the PBX. > > And if you took out the asymmetric routing, the other symptoms would make > me > think it's a NAT traversal or SIP ALG problem in the customer's router. > But > with the dependency on how the traffic flows over the Internet, I haven't a > clue. > > > -----Original Message----- > From: AF <[email protected]> On Behalf Of Nate Burke > Sent: Thursday, December 3, 2020 5:18 PM > To: Animal Farm <[email protected]> > Subject: [AFMUG] Offnet SIP and Multi-homing > > I'm trying to track down a strange Issue I'm having, and wondering if > anyone > has run into something similar. > > I have a couple customers that I let take phone handsets home, the > Grandstream PBX is at our office. It seems that if the SIP traffic comes > in > one upstream, but leaves another upstream, really weird things happen. > > The Phone registers to the PBX just fine, and can receive calls all day > long, it just cannot make calls. If I take a /24 and force it to enter and > leave the network via the same provider, then everything works fine. Those > of you that have multiple diverse paths, have you seen something like this? > The Call in UDP Mode fails, the call in TCP mode will succeed. > > This only appears to be an issue with the handset talking to the PBX. > All of our Sip Trunk traffic (Voip Innovations) works just fine. For some > reason the handset SIP re-invite never makes it back to the handset when > multi-homing is in place. > > > > -- > AF mailing list > [email protected] > http://af.afmug.com/mailman/listinfo/af_af.afmug.com > > > > -- > AF mailing list > [email protected] > http://af.afmug.com/mailman/listinfo/af_af.afmug.com > > > -- > AF mailing list > [email protected] > http://af.afmug.com/mailman/listinfo/af_af.afmug.com > -- Lewis Bergman 325-439-0533 Cell
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