SIP Trunk between two CUCM Using NAT

[Cheyenne.Belisario]

I made a SIP Trunk between two CUCM Nating the Cucm Ip Address with a Public IP on both sides, on my ASA firewall. I opened all the necessary ports.

The call works, the Phone is ringing but I do not have Audio. 

[Georgios Fotiadis]

Have you turned on SIP inspection on the firewalls? Which ports exactly have you opened? Also, beware that if you do not have MTP on CUCM then RTP will flow directly between the IP phones.

[Cheyenne.Belisario]

Hi Georgios,

Yes I have Turned on SIP inspection on my firewall. in this moment I opened all ports making test. on SIP trunk I do not have Available the MTP OPtion. 

[Georgios Fotiadis]

Are your phone subnets also NATed? If you go to the phone GUI, under the streams section, you will be able to see the IP addresses between which RTP packets flow. Look for the active stream on the phone GUI.

[Cheyenne.Belisario]

My phone subnet is does not Natted, I only Natted the CUCM ip Addres. 

[Georgios Fotiadis · Accepted solution]

After call establishment, RTP flows between the IP phones directly; i.e. they should be able to communicate with one another over the WAN. Your options:

a) NAT phone subnets

b) Create a site-to-site VPN and allow CUCM/phone traffic through it

c) Enable MTP on SIP trunks. You will need Media Resources here either on CUCM (not recommended), or an ISR. But that will enable MTP for all calls; which will "cost" you in DSPs.

Personally, I would chose (b). For testing for can do either (a) or (c).

[Cheyenne.Belisario]

I also Prefer option B but I going to trying with the MTP. Thnaks for you help