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No speech path on BCM to BCM via the Internet using H323 or even via the net with an IP Phone 1

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Firebird Scrambler

Systems Engineer
Apr 7, 2003
7,205
GB
Hello everyone

I'm not sure if anyone reading has ever successfully done this, but I've seen posts on this topic before over the years and it seems that it's not possible for one reason or another?.

Although I can set up a H323 call and also register an IP Phone OK via the internet, I can't ever get any dial tone or speech path.

Before I go into details, I want to make it known that I have two BCM 50 systems with 2 V.O.I.P. H323 trunks and IP Licences and everything works fine when both systems are connected via my internal network.

i.e. BCM Main = 86.130.12.5 and BCM test = 86.130.12.6 etc.

I have two separate broadband internet routers with different ISP Providers and I've been through the port settings for the routers to give them the same settings under "NAT" on each one in "Port Forwarding" such as

BCM Call Pilot & EM 443 443 443 443 86.130.12.5
BCM Trunk Signalling 1718 1720 1718 1720 86.130.12.5
QoS Probe / SIP 5000 5060 5000 5060 86.130.12.5
BCM Call Pilot & EM 5989 6001 5989 6001 86.130.12.5
BCM Signalling 7000 7002 7000 7002 86.130.12.5
BCM Signalling 16500 16501 16500 16501 86.130.12.5
I20xx to BCM 28000 28511 28000 28511 86.130.12.5
BCM Monitor 60001 60001 60001 60001 86.130.12.5
I20xx to BCM 51000 51063 51000 51063 86.130.12.5

Although a number of the ports mentioned above aren't required for IP and Trunk access, I can however access my BCM using Element Manager and the Monitor tool OK via the internet.

From reading all the notes on port settings I'm told that

UDP 1718 to 1719 Used by H323 protocol
TCP 1720 Used by H323 protocol
UDP 7000 Used only when IP sets are active. Signaling to sets takes place on this port.
UDP 7002 Used to handle UNIStim-based IP set firmware file transfers for the purpose of managing IP set F/W
UDP 28000 to 28511 Voice Path for IP trunks, adjustable via Element Manager. See also ports 1718-1720
UDP 51000 to 51063 Voice Path from RTP/RTCP Streaming (intercom/gateway dial tone)

I've even changed the H323 ports 1718 to 1720 to be both UDP and TCP.

With regards to using an IP phone via the internet, I've tried many times, but still unable to get any dial tone after the phone has registered.

I do however get a major alarm 50115 appear which is
TERM.(0004, :0001): No public media address available for IP set at 86.130.12.49:51008

It suggests I "Contact your local support group."!

This error has appeared before, but the port 51008 has already been defined in "Port Forwarding" table range on 51000 to 51063 and I've even made that range UDP and TCP with no effect!.

Alarm 51005 appears every time I power up an IP phone. I've used a 2050 and 2004 phone with the same results from both systems.

I even set up each router with all ports to point via the Demilitarized Zone (DMZ) to the BCM 50 in case there was a port that I was missing?. This made no difference.

I've got to the point of wondering if the problem with I think has to be the media or speech path is due to some sort of port mapping translation?. Its as if the speech or dial tone is received or sent via different ports from each end. Has anyone come across this before within a network and was it easily resolved?.

I've also tried connecting to another BCM in a different country and I still get the same problems at both ends.

It's got me stumped and another thought was to use some sort of "easy to use" network sniffing tool so that it easily shows what is happening.

Thanks for taking the time to read all of this as it's taken me 2 to 3 hours to write it all down!.




All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

If it's working, then leave it alone!.
 
Check the SIP ALG (Application Level Gateway) setting in your Router. I realize you are doing H.323 but the SIP ALG can block speech path on the ports that are used.
From a Linksys\Cisco Router-"SIP ALG

The SIP ALG option allows SIP packets to traverse the NAT firewall. To use the SIP ALG for VoIP service, select Enabled. If your VoIP service provider uses other NAT traversal solutions such as STUN, TURN, or ICE, then keep the default, DISABLED.

Also consider Port Forwarding Ports Range 10,000-13,999 to the BCM with UDP protocol. We do this when using UCx and Nortel sets as well.
Just a thought for you....

Stocking Distributor for E-Metrotel
 
I forgot to add that I'd rather not put here my published IP address for each BCM, but feel free to contact me via the link below if you want them to carry out your own testing.

All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

If it's working, then leave it alone!.
 
Hello Guys

I've been into both routers and ticked the SIP ALG one as it was disabled before.

Enable or Disable SIP ALG

ALG State
ALG :
Enable Disable

I was only able to find one router with this setting.

Also BCM IP Signalling 10000 13999 10000 13999 86.130.12.5 UDP
has been added in on both routers.

I've even turned off the firewall on both routers at the same time and still I can't get dial tone or speech working.

And I've made H323 calls as well as connecting one of the IP phones via the internet and still can't get the dial tone when I press the intercom key.






All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

If it's working, then leave it alone!.
 
We have had to turn SIP ALG OFF for proper operation. This looks like some sort of port issue. If you can, try registering the IP phone to a SIP service like voip.ms and see if it will there. One last thing, since this is over Internet and not a WAN, try turning the QoS settings off since QoS can't be guaranteed over the Internet.

Stocking Distributor for E-Metrotel
 
It now looks as if I might have sorted out the H323 BCM to BCM problem!.

A search via a tip from my good friend Curlycord with a reference to

I saw this below.

The first digit of the extension range of the remote site.
Destination IP: IP address of the remote site.
GW type: BCM
GW Protocol: CSE
Voip Protocol: SIP ############ bingo! ###########
Check QoS monitor checkbox.

All this time, I had my settings set on both BCM systems to H323 because my trunks were H323. Once I had set them to SIP, I was getting ringing tone from the calling end and speech at both ends once connected.

I mad sure the GW Protocol was set at CSE, but it didn't matter about the QOS setting being ticked or not.

All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

If it's working, then leave it alone!.
 
Nice!

=----(((((((((()----=
curlycord

small-logo-sig.png

Toronto Canada
 
Further testing has confirmed 1 way transmission and that is because of the "SIP ALG" setting which as you say, needs to be enabled. Just got to find it on my other router!. Looks like it's not there in the configuration.

As for the I2004 phone, I still can't get any dial tone when I press the intercom key after I have registered it via the internet.

All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

If it's working, then leave it alone!.
 
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