Telepresence cannot make a video call locally when MTP required is enabled

[mitch_santos_21]

Hi Everyone,

I have a problem with EX90 and EX60 Telepresence, here are the scenarios:

1. When telepresence is making a video/audio calls to another telepresence locally, it is ok and succesful. (Take note: MTP required is not checked)

2. When telepresence is making an audio calls to both IDD and mobile calls, it is not succesful. You need to enter a pin in order to make an IDD and mobile calls. However when we are trying to enter the pin, a voice prompt is heard "unable to authorized". In order to isolate it, we make an audio call using an IP Phone and the calls to IDD and mobile calls are succesful using the pin we entered when we are trying to make a call using telepresence. When I checked the phone configuration of telepresence, MTP required is not checked.

3. When we checked the MTP required, Audio Calls from Telepresence to IDD/Mobile Calls are ok, but the video call from telepresence to telepresence locally are not ok anymore. Audio calls are both fine from telepresence to IDD/mobile calls and Telepresence locally.

Hope you can assist us on this problem.

Please see the attached files also.

Thanks,

[Jonathan Schulenberg]

The "MTP Required" checkbox is usually a workaround, not a solution. In this case it likely allowed DTMF interop between the RFC2833 that the EX90 uses and whatever your gateway is expecting (e.g. MGCP out-of-band). The reason you're having problems with video and the MTP is that the MTP doesn't support more than three RTP channels until very recently (exact code version escapes me); thus video can't negotiate since it requires at least four.

What's the PSTN gateway configuration? Focus on getting it to support RFC2833 DTMF.

Please remember to rate helpful responses and identify helpful or correct answers.

[mitch_santos_21]

Hi,

Please be advised that the protocol used for EX90 is SIP. How do we know if the MTP use supports more than 3 RTP channels?

What do you mean by "focus on getting it to support RFC2833 DTMF"? Can you please elaborate it? Does it mean we need an RFC2833 DTMF compliant MTP device?

Do we need to make some configurations in voice gateway and CUCM?

By the way the call flow will be like this:

When calling IDD/Mobile

Telepresence --> CUCM--> ACS (Firewall) --- h323 --> VGW

Regards,

Mich

[Alok Jaiswal]

Hi Michell,

check this thread, is this created by your side only ?

https://supportforums.cisco.com/message/3766269#3766269

As mentoned by jonathan the MTP required could be a workaround but not the solution. After you enable this option the CUCM will try to invoke MTP resource for the call.

video codec are not supported by CUCM soft MTP and for hard MTP's you should have new PVDM-3 dsp resources on the router. PVDM-2 doesn't support video codecs.

Can you tell me what it your gateway type? MGCP/H.323/SIP??

Thnx

Alok

[mitch_santos_21]

Hi Alok,

The gateway type is H323.

Regards,

Mich

[Jonathan Schulenberg]

Set this on your VOIP dial-peers and try again. Make absolutely sure that the call is matching those dial-peers and not dial-peer 0!

dtmf-relay h245-alphanumeric rtp-nte

Please remember to rate helpful responses and identify helpful or correct answers.

[mitch_santos_21]

In RFC 2833 and OOB method for DTMF signaling, do we need to made some configuration in both voice gateway and CUCM?

Thanks and regards,

Mich

[Alok Jaiswal]

Hi Michell,

yes you need to configure the command on to the necessary dial-peers in voice gateways.

rgds,

Alok

[mitch_santos_21]

Hi,

I already tried to add that config, however voice prompt of "unable to authorized...."

here is the dial peer config.

dial-peer voice 21 voip

preference 1

destination-pattern ....

session target ipv4:10.0.220.1

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 22 voip

preference 1

destination-pattern ....

session target ipv4:10.0.224.1

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs5 media

no vad

!

dial-peer voice 3 pots

preference 6

service pin

destination-pattern ....

direct-inward-dial

forward-digits all

!

dial-peer voice 4 pots

preference 6

service pin

destination-pattern ......

direct-inward-dial

port 3/0:0

forward-digits all

!

dial-peer voice 69 voip

service pin

incoming called-number 90

dtmf-relay h245-alphanumeric rtp-nte

codec g711ulaw

!

dial-peer voice 70 voip

service pin

incoming called-number 91

dtmf-relay h245-alphanumeric

codec g711ulaw dial-peer voice 21 voip
preference 1
destination-pattern ....
session target ipv4:10.0.220.1
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!
dial-peer voice 22 voip
preference 1
destination-pattern ....
session target ipv4:10.0.224.1
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!
dial-peer voice 3 pots
preference 6
service pin
destination-pattern ....
direct-inward-dial
forward-digits all
!
dial-peer voice 4 pots
preference 6
service pin
destination-pattern ......
direct-inward-dial
port 3/0:0
forward-digits all
!
dial-peer voice 69 voip
service pin
incoming called-number 90
dtmf-relay h245-alphanumeric rtp-nte
codec g711ulaw
!
dial-peer voice 70 voip
service pin
incoming called-number 91
dtmf-relay h245-alphanumeric
codec g711ulaw

regards,

Mich

[mitch_santos_21]

Hi,

I was able to solve this problem with the help of Cisco TAC. The resolution we made is creating a SIP trunk for all IDD/mobile calls in CUCM and creating a SIP dial peer in voice gateway. The DTMF signalling method on the trunk is rfc2833. The dtmf relay in voice gateway is rtp-nte.

Best regards,

Mich