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 <af-boun...@af.afmug.com> On Behalf Of Nate Burke
Sent: Thursday, December 3, 2020 5:18 PM
To: Animal Farm <af@af.afmug.com>
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.
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