Dear Expert,
Mentioned below is the topology which I am trying to put online but struggling to make it work.
At this time working on the 1st step only i.e making outbound call to PSTN end point ;-)
Call Flow:
1)
SIP Tandberg Video End point --> Tandberg VCS------> SIP Trunk -----> CUCM8.5.1 ---->SIPv2 Trunk ----> ISR2951(15.2)---->PRI Link----PSTN Video End Point
Result = At the ISR GW : 6 Channels getting bonded, call connects, No Audio and Video
In the ISR GW: Call get stuck at
##################################
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.250.20.22
s=SIP Call
c=IN IP4 10.254.212.21
t=0 0
m=audio 46116 RTP/AVP 0 102
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-15
m=video 46118 RTP/AVP 121
b=TIAS:1920000
a=rtpmap:121 H264/90000
a=fmtp:121 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=395500
a=inactive
a=rtcp-fb:* nack pli
rtcp-fb parse payload numtok not foundrtcp-fb payload found, specific is 25
########################################
2)
H323 Tandberg Video End point --> Tandberg VCS------> SIP Trunk -----> CUCM8.5.1 ---->SIPv2 Trunk ----> ISR2951(15.2)---->PRI Link----PSTN Video End Point
Result : Call get connected, 6 channels get bonded at the ISR GW, The Video end point making call to PSTN only Tx Audio and Video but for Rx there is no audio and video. Video end point GUI status shows "Place on Hold"
Please note following configuration specific to each device :
-the SIP Trunk between VCS(7.1) and CUCM- used the latest guideline
-SIP Trunk between CUCM and ISR - basic config using the default SIP profile
- At CUCM, Both SIP trunk use the same MRGL,Location and Devicepool
-At the ISR GW Sample Config(ISR Gw has PVDM3):
#########################
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
emptycapability
no h225 timeout keepalive
h245 timeout tcs 40
modem passthrough nse codec g711ulaw
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g722-64
video codec h263
video codec h263+
video codec h264
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn timer T310 60000
isdn bchan-number-order ascending
isdn sending-complete
isdn integrate calltype all
no cdp enable
!
voice-port 0/0/0:23
voice-class called-number-pool 1
!
dial-peer voice 300 pots
description OUTBOUND
destination-pattern 91[2-9]..[2-9]......
progress_ind setup enable 3
progress_ind progress enable 8
information-type video
bandwidth maximum 384
direct-inward-dial
port 0/0/0:23
forward-digits 11
!
dial-peer voice 20 voip
description INBOUND VOIP
rtp payload-type cisco-codec-fax-ack 102
rtp payload-type cisco-codec-video-h264 97
session protocol sipv2
incoming called-number .
voice-class codec 1
voice-class sip calltype-video
dtmf-relay h245-signal h245-alphanumeric cisco-rtp rtp-nte
ip qos dscp cs3 signaling
no vad
!
################
Thanks in advance for your kind attention and hope to hear from you soon.
Cheers
