sip trunk, h323 gateway and outgoing call in cucm

[s.maxina1]

Hi dears.

i have these conditions in my office:

Head Quarter: CUCM 10.5 + Cisco 3945 + E1 Modem >>>>>>to PSTN

Branch A: phone A connected to FXS port of third party Voip gateway(New rock brand) integrate with CUCM using sip trunk.

Branch B: phone B connected to FXS port of third party Voip gateway(New rock brand) integrate with CUCM using sip trunk.

Branch C: phone C connected to third party Voip gateway integrate with CUCM using h323 Gateway.

if phone A call phone B, based on previous question, i unchecked "Media Termination Point Required". because i don't want media coming back to Head Quarter( after call setup i need call from branch A to B directly).

in this case by unchecking "Media Termination Point Required", 2 problems occur. first one, phone A can not call PSTN (for example mobile phone line). another one is that, when phone A, call Phone C, after off hooking phone C, call will be dropped. 

how can i solve these problems while "Media Termination Point Required" is checked in sip trunk?

Best regards.

[Nithin Eluvathingal]

You cannot Uncheck "Media Termination Point Required" for a reason that you  don't want media coming back to Head Quarter.

Our reply doesn't mean that when ever you need traffic to Flow   between endpoints, uncheck the Media termination Point required option. 

[s.maxina1]

Thanks nithin. I have to explain more. Branch A and Branch B are linked to head quarter via VSAT( satellite communication) which has high latancy about 600 ms( totally 1200ms very very high). Head quarter also linked to VSAT provider company via point-to-point wireless link. With this topology, it is prefered to uncheck mtp. If i wrong, pls correct me.

Best regards

[Roger Kallberg]

With this topology you would never get a good working solution as the delay is well past what is within the boundary for what is recommended for a voice over IP solution. Recommend you to read the appropriate sections in the SRND to get more information on this.

[s.maxina1]

Hi roger. Branch A and B are location of onshore drilling rigs in desert without any communication such as 3g/4g or ptp wireless link except VAST. 

[Nithin Eluvathingal]

 MTP in  a telephony deployment is varied and diverse depending on your design, components and technology. Disabling MTP option on the trunk may not be a solution for you.

You Latency is too high and its not asper the Voip recommendations. Try QOS if possible on VSAT or You may need to think of decentralized design,  or an alternative connectivity options

[s.maxina1]

This is the only solution right now and when I unchecked MTP, baesd on the report of local IT MAN, better call experience is achieved, but outgoing call to pstn and to branch C was dropped. 

[Nithin Eluvathingal]

As i mentioned MTP in a telephony deployment is varied and diverse depending on your design, components and technology. 

Regarding your Issue Outgoing call to PSTN failed.

• CUCM 10.5 + Cisco 3945>> what gateway protocol your use for 3945
• From which trunks, you removed the MTP
• How many trunks you have in your deployment
• Since you are suing E1 to connect PSTN, please collect debug isdn q931 debugs by recreating the issue.

Regarding branch C  call drop.

• when phone A, call Phone C, after off hooking phone C, call will be dropped.>>> what Codec you use for this calls ?
• phone C connected to third party Voip gateway integrate with CUCM using h323 Gateway.>> what changes you made on this gateway ?