Pmp450 and SIP registrations

Dear Sirs,

what is your experience with voip sip registrations behind PMP450?

Our customers complaints that they are not able to have their SIP devices registered with external servers (with 14.1.1 fw too)

With or without port forwarded and with or without nat traversal options enabled on the external registrar, the devices don't register and the NAT type seems symmetric (the worst for voip).

We have to put the SM in bridge mode by adding a router for the autenthication and then the voip device work.

With epmp1000 no problems, since they have the nat helper ALG functions too.

Kind regards

Rocco

COMeSER

Hi Rocco,

What is the software version on which the cnPilot is running on.

If the device is getting registered behind the ePMP1000 and with this set up, if possible can you please make a call to the other phone and share us the packet trace from the cnPilot and the step is  mentioned below to perform the same.

1st packet trace from ePMP1000

  1. Start packet trace
  2. Dial registered R200 from another registered R200
  3. Call will get connected and then you can wait for some 10 to 20 seconds
  4. Hang up
  5. Stop packet trace 

2nd packet trace from  PMP 450

  1. Start packet trace
  2. Dial unregistered R200 from another unregistered R200
  3. Call will not get connected so wait for 10 to 20 seconds
  4. Hang up
  5. Stop packet trace 

Once you connect the cnPilot behind the PMP450 kindly try pinging the SIP server and see if it is getting reached to the server.

Thank You.

Regards,

Yogesh Giriappa

Cambium Technical Support

Hi,

the voip devices are not CnPilot R200 but generic, self installed devices from customers.

When we give OUR voip service we use R200 or Patton devices and the pppoe authentication is made on the same device acting as voip gateway (so no NAT performed on the SM since it is in bridge).

For those customers and their self installed voip solutions, the pmp450 SM get its IP and does NAT.

Their external sip server has Nat traversal techniques and port forwarding are not requested. The only requisite is that the NATing router (the SM in their case) is not from symmetric but full cone.

Kind regards

Rocco

Are the customers running their SIP client behind their own router (NAT) behind the SM (more NAT)?

You might try a DMZ to their own router or ATA.

Hello,
I have the same problem as "roccoptr".


They are the customers who use their own SIP device behind a NAT configured SM mode.

Is there any solution to this problem, some extra configuration on the AP or SM?

What should we do with these customers who already have their own SIP solution and are migrating to Cambium to provide better quality service?

Thanks a lot.

Hope this is ok, I know this is a much older post but found this and wanted to inquire.  I have company next door to us that has voip phones from their old ISP out of town.  So from their office they go through us via a NAT enabled PMP 450 to our tower which then goes to another ISP that carries our Fiber. This then goes to another ISP that is hosting their viop phones. 

I have read about voip settings in the pmp450 and have set up the CIR in the QOS and the DIFFSERV 46 code to +7 in both the AP and SM. They have 5 phones so in the low priority feild I put their package plan and in the High Priority I put 1000kbps giving them 200kps per phone, which should be enough I would assume. I also put the QOS bursting settings to 0 ( tried to hone those in and same result) So the phones give this client their internet, this was against our advice but we just brought them internet they already had this set up before moving up here with this other company. 

Now they get their phones and internet flowing about 5-20 minutes before they say they lose it all. We see more of the ~5 min mark before they fail. You can make calls and recieve calls but it drops every time. They have another land line internet there for another company they house with and have had zero issues. 

    The one company they use is different than our fiber provider too. 

The other parts of this lol, sorry for the book. 

At my Desk I am on my network, same signal and everything that they have next door. ( I am static bridged and they are nat) I have a voip phone and have never had an issue with it dropping or erroring out. We brought their phone to my office and I plugged it in and it acted like it had no internet at all. I feel its a configuration issue with voip or our upstream but wanted to be sure I was not missing something, again shocked it did not work at my office, as they said these voip phones can be put in anywhere and work, so again not sure. We are trying to reach out to the voip providers but they have been saying its just our network cause it works on someone elses. We do not wanna lose this client so really would like to solve this issue. Figured it was worth a shot to throw this in here and see what happens. 


@Fullspeed wrote:

Now they get their phones and internet flowing about 5-20 minutes before they say they lose it all. We see more of the ~5 min mark before they fail. You can make calls and recieve calls but it drops every time. They have another land line internet there for another company they house with and have had zero issues. 

...

At my Desk I am on my network, same signal and everything that they have next door. ( I am static bridged and they are nat) I have a voip phone and have never had an issue with it dropping or erroring out.


Can you clarify exactly what happens when you say "drops"?

In general it sounds like a very classic NAT problem, possible flow/state-tracking resources on the SM are being exhausted.