I came across your blog when doing some research and its totally great. I have been struggling with getting CUVA to work with our deskphones.
To give you a quick background, we are running UCCE version 7.5. Our Desk phones are 7941’s with the latest firmware (9-1-1SR1S). All deskphones are configured as SIP.
Im running Video Advantage verion 2.2 (which supports SIP video) and cannot get the phone to integrate with the video software. CUVA works perfectly fine with a deskphone. One theory I had was the windows firewall was blocking CUVA from speaking with the 7941 but after disabling it, Im still experiencing the issue.
Is there soemthing major that Im missing, which is causing the webcam to no communicate with the phone?
Just as a reminder when you sit in the lab, try to not forget to put the correct Calling number , type and plan and this for all situation so it is true as well for SRST.
So a good vision of the call routing is important
Real-time fax over IP operates in a similar way of a regular fax transmission. The fax machines involved in the transmission synch up and then the fax data is sent between them over the intervening IP Network.
There are 2 methods of transporting fax in real time across the network:
When using fax-relay, the T30 fax signal from a connected fax machine is demodulated by the sending fax gateway and sent over the IP Network to a remote fax gateway. The remote fax gateway then recontructs the T30 fax signal and send it to the fax.
There are 2 types of fax-relay mechanisms:
Cisco fax-relay is an older method. So a fax gateway terminates T30 fax tones from a local fax machine and then sends the fax data across an IP network by breaking the tones into HDLC frames and then transmitting them using RTP.
T38 fax-relay is the ITU standard T30 fax signal, it is demodulated at the local gateway and encapsulated into IP packets for transport over a network to a remote fax gateway which will then reconstruct the signal and play it to the fax. T38 includes also a mechanism by which a fax gateway can inform the remote gateway of its desire to change the media type from voice to data. T38 can also use TCP or UDP connections but will use more UDP.
For fax pass-through, modulated fax data is sent in-band across the IP network by a fax gateway using a voice codec (like G711 without any VAD or echo-cancellation). Also with fax pass-through, T30 fax calls are not distinguished from regular voice calls, they are simply sent in-band over the IP Network. With the fax detection tones, the gateway must be able then to switch to high-bandwidth codec. Fax pass-through is relatively bandwidth hungry and is sensitive to delay,jitter and packet loss
Here are all the steps to perform if you want to integrate the CUE with the CME
- Create a SIP dial peer pointing to the CUE Module
- Configure the MWI On/Off extensions
- Configure the connectivity between CME and CUE Modules
- Perform CUE Configuration
For the first step , your dial peer will look like
dial-peer voice 1000 voip
session protocol sipv2
session target ipv4:172.17.1.1
The configurations of the MWI On/Off must be generic as it must cover all numbers and it is generally implemented as a kind of prefix as the MWI must know which number must be turned on or off. So it adds the number after the extension the MWI shortcut.
Regarding the link configuration between the CUE and CME, you need to borrow the Ethernet interface where the CUE module resides (don’t forget that CME and CUE can be splitted). So everything under the service-engine ( this is the CUE module) will rely on the Ethernet interfaces:
interface Service-Engine 1/0
ip unnumbered FastEthernet 1/0
service-module ip address 172.17.1.1 255.255.255.0
service-module ip default-gateway 172.17.1.254
ip route 172.17.1.1 255.255.255.255 Service-Engine1/0
In a multi-site implementations , you have then to configure a translation between your remote CME and the central CUE/CME because we told before that CUE relies on SIP Protocol and the rest of your network is an H323 network , so you need to translate all H323 request to SIP. You can do it generally with the config:
voice service voip
allow-connections H323 to sip