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Sergey Balyura

Video over sip trunk from CME to CUCM and back

Hello everybody!

I need help so much!

I have CME (IOS 15.1(4)M2 and CME 8.6) and CUCM 7.1

I have third-party SIP client on CME - and 9951 on CUCM.

I try to make a video call. but still cannot.

every call maiden is audio only.

inside each site - inside CUCM or inside CME - everything is working fine with video

here is a part of my CME config:

voice service voip

no ip address trusted authenticate

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

fax protocol none

modem passthrough nse codec g711ulaw


  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  registrar server

  asymmetric payload full

  pass-thru content sdp

voice class codec 4101

codec preference 1 g711ulaw

codec preference 2 g711alaw

video codec h261

video codec h263

video codec h263+

video codec h264

video codec mpeg4

voice register global

mode cme

source-address 192.168.AB.CDE port 5060

max-dn 30

max-pool 30

authenticate realm all

timezone 32

tftp-path flash:

file text

create profile sync 031109583450814A



voice register dn  2

number 188

voice register pool  2

id mac 0000.0000.0000

number 2 dn 2

dtmf-relay rtp-nte

username 188 password 188188

codec g711ulaw

no vad



sccp local GigabitEthernet0/1

sccp ccm 192.168.AB.CDE identifier 1 version 7.0



sccp ccm group 1

bind interface GigabitEthernet0/1

associate ccm 1 priority 1

associate profile 1 register XCODE

keepalive retries 5

switchover method immediate

switchback method immediate

switchback interval 15

dspfarm profile 1 transcode 

codec g729abr8

codec g729ar8

codec g711alaw

codec g711ulaw

codec g729r8

maximum sessions 4

associate application SCCP

dial-peer voice 201 voip

translation-profile outgoing TO_OFFICE

destination-pattern 700...

session protocol sipv2

session target ipv4:192.168.KL.MNO

voice-class codec 4101 

no voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

no vad 


sdspfarm units 5

sdspfarm transcode sessions 10

sdspfarm tag 1 XCODE

no auto-reg-ephone

max-ephones 30

max-dn 30

ip source-address 192.168.AB.CDE port 2000

max-conferences 8 gain -6

transfer-system full-blind

transfer-pattern .T

create cnf-files version-stamp 7960 Feb 24 2014 12:00:01

on CUCM I have SIP trunk with DTMF - RFS 2833

So where am I wrong? please help!!!

Sergey Balyura

anybody? please help!

A bit late, but try this:

dial-peer voice 201 voip

 voice-class sip pass-thru content sdp

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