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Does this Topology works or in use for Video and Audio call ? Urgent

Unanswered Question
Aug 18th, 2012
User Badges:

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

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