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syadnom
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SIP Helper

Mon May 11, 2015 9:59 pm

I'm having some troubles getting SIP helper to work.

I'm running grandstream GXW4004 gateways to Voip Innovations.
I feel like I've tried every possible config.

port 5080 for sip, 5090-5094 for RTP. No registration is used.

Basically, I want to avoid entering in the NAT IP address in the grandstream, and instead have SIP Helper adjust the 'src' address. I have a redundant internet connection and when the gateway switches to the backup, I want the SIP Helper to write the new gateway because the grandstream doesn't have a mechanism to change this.

Also note that the grandstream's STUN settings don't update the return address so STUN can't be used to solve this.
 
barkas
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SIP Helper

Mon May 11, 2015 10:20 pm

Usually, always disable the sip helper.
 
syadnom
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Re: SIP Helper

Mon May 11, 2015 10:21 pm

In this circumstance, I need either sip helper, or I'll have to add a sip proxy
 
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ZeroByte
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Re: SIP Helper

Tue May 12, 2015 1:37 am

I inherited several dozen sites that had Mikrotik w/ SIP helper + Grandstream gateways behind it.

All I can say is MAJOR NIGHTMARE!

Any site where we disabled the helper would immediately begin working MUCH better.
(this was how I even found out that there is a SIP helper in Mikrotik in the first place!)

The SIP provider almost always has an SBC which is smart enough to properly detect NAT and correct for it automatically.
Also make sure the SIP keepalive interval is short enough to stop the UDP connection tracking from timing out in the Mikrotik. If the GS thinks it's still registered and the server thinks the GS is dead, then only rebooting the GS (sometimes the GS AND the Mikrotik) ever fixed that scenario.
When given a spoon,
you should not cling to your fork.
The soup will get cold.
 
syadnom
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Re: SIP Helper

Tue May 12, 2015 1:43 am

No SBC NAT detection available w/ VoIP Innovations. Additionally, there is no registration, calls are sent to the site's IP for termination and only calls from that same pre-defined IP address(es) are accepted for origination.

The SIP helper does work for inbound. With it off and no NAT IP defined it never connects, with it on all is well. Outbound calling on the other-hand doesn't work with it on.

I've been arguing with grandstream support that the entire purpose of STUN is to identify the WAN address and set that as the contact address in the SIP session. They've offerred various dumb solutions (dynamic DNS, using two profiles and manually changing the profile) which makes their STUN setting useless :/
 
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ZeroByte
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Re: SIP Helper

Tue May 12, 2015 1:54 am

Sounds like a static IP from your ISP would make your pain go away.
When given a spoon,
you should not cling to your fork.
The soup will get cold.
 
syadnom
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Re: SIP Helper

Tue May 12, 2015 1:55 am

I have static IP addresses from both ISPs.
 
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ZeroByte
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Re: SIP Helper

Tue May 12, 2015 2:12 am

I have static IP addresses from both ISPs.
So you don't have a static IP because you're doing failover..... hence the reason you can't put the NAT IP in the Grandstream. . . I see the problem for sure...

Well, all I can say is that in my personal experience, the SIP helper in Mikrotik caused Grandstream to break in unpredictable ways at unpredictable times, usually quickly fixed by rebooting the Grandstream, which is NOT a solution.

Our solution was to disable SIP ALG which fixed 99% of the sites, and for a couple of stubborn ones, we had to route IPs directly to them and disable NAT entirely. I wish I had better news for you from my experience with this gear.

Granted, our switch had an SBC which also did the NAT traversal, and two devices working at the same time to do NAT traversal tends to cause issues. Is it possible for you to use the standard SIP ports?

If not, did you set the service ports to the ones your provider is using, since 5060,5061 are the standard?

If you're using PCC, you might exempt the Grandstream / SIP provider from PCC and use primary/failover mechanism.
If the RTP happens to get mapped to the opposite ISP, then that could cause issues...
When given a spoon,
you should not cling to your fork.
The soup will get cold.
 
syadnom
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Re: SIP Helper

Tue May 12, 2015 10:30 pm

I *could* use standard ports, but 5060 is constantly port scanned so I don't like to use it. The provide does direct media so I can't just firewall off their SIP port else I block RTP and unfortunately, that RTP stream doesn't come in as a 'related to' the SIP connection :(

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