09-06-2010 06:02 PM - edited 03-17-2019 10:05 PM
We currently have our Cisco Telepresence solution working well via a CUBE to a Tandberg Telepresence server, but we are unable to get a Cisco 7985 working via the CUBE. A CCSIP debug shows a "no sdp" error. This may have something to do with having "sdp passthru" enabled on the CUBE (?) for Telepresence. The SIP trunk from CUCM is just a basic default configuraiton.
The dial peers are also just basic configuaration (no codecs specified) - I've tried with "codec transperant" both on and off on the dial peers.
Any ideas?
09-06-2010 07:20 PM
I should clarify that it works "audio only"
09-09-2010 10:43 AM
This is a standard deployment for it to work correctly you need to invoke the invia zone i.e. cube and have the following commands
h225 connect-passthru
call start slow
no call sync-rsvp slow-start
h245 tunnel disable
h245 caps mode restricted
h245 passthru all
voice class codec 1
codec preference 1 transparent
voice class media 2
media flow-through
dial-peer voice 2 voip
description *** IP/IP ***
huntstop
voice-class media 2
voice-class codec 1
session target ras
dtmf-relay h245-alphanumeric
no vad
So Usually this config works if it doesnt then send me the following trace and i wil let you know why the negotiation is failing.
deb cch323 all
deb voip ccapi def
deb h225 asn1
deb h225 eve
deb h245 eve
deb ip tcp transaction
deb h245 asn1
deb voip ipip
Thanks
09-12-2010 03:01 PM
With "media flow through" enabled does that mean the CUBE won't be proxying the media and/or performing address hiding? We need it to act as a proxy between two networks.
Either way I will be trying out the config you've provided - thanks very much.
09-21-2010 05:44 PM
Can it work without using a gatekeeper - i.e. straight sip-to-sip as per Telepresence?
Currently, we dial into the Tandbeg TP server with a 7985 (using SIP), it connects, then says "Attention: no incoming video data" and disconnects after approx. 10 seconds.
The config we have below works perfectly for Telepresence. I tried removeing the rtp payload types from the dial peers as well as specifying the video and voice codecs (h264 and g77ulaw) but the result was the same. I also added huntstop, no vad and the voice classes as specified in the above config to the dial peers, but same result.
Config:
!
voice service voip
rtp-ssrc multiplex
address-hiding
allow-connections sip to sip
sip
session transport tcp
rel1xx disable
header-passing error-passthru
no update-callerid
midcall-signaling passthru
pass-thru content sdp
!
!
!
!
!
!
!
!
!
!
!
ccm-manager fax protocol cisco
!
mgcp fax t38 ecm
!
!
!
dial-peer voice 1 voip
description dial peer to Callmanager
rtp payload-type cisco-codec-fax-ind 110
rtp payload-type cisco-codec-aacld 96
rtp payload-type cisco-codec-video-h264 112
session protocol sipv2
incoming called-number 14...
destination-pattern 14...
dtmf-relay rtp-nte
codec transparent
session target ipv4: [Callmanager IP]
!
dial-peer voice 2 voip
description dial peer to Tandberg TP server
destination-pattern 10..
rtp payload-type cisco-codec-fax-ind 120
rtp payload-type cisco-codec-aacld 96
rtp payload-type cisco-codec-video-h264 112
session protocol sipv2
incoming called-number 10..
session target ipv4:[Telepresence Server IP]
dtmf-relay rtp-nte
codec transparent
!
!!
gateway
timer receive-rtp 1200
09-23-2010 04:27 PM
I should add that there is a Tandberg VCS between the CUBE and the Tandberg Telepresence Server.
So the call flow is 7985 -> CUCM -> CUBE -> VCS ->Telepresence Server
We managed to get the exact same result with the result with the config below, which is basically the same minus the Telepresence specific config - i.e. connects for a few seconds, says "no incoming video" and disconnects:
voice service voip
address-hiding
allow-connections sip to sip
sip
pass-thru content sdp!
!
!
!
!
!
!
!
!
!
!
!!
!
dial-peer voice 1 voip
description dial peer to Callmanager
session protocol sipv2incoming called-number 14...
destination-pattern 14...
dtmf-relay rtp-nte
codec transparent
session target ipv4: [Callmanager IP]
huntstop
no vad
!
dial-peer voice 2 voip
description dial peer to Tandberg TP server
destination-pattern 10..
session protocol sipv2incoming called-number 10..
session target ipv4:[Telepresence Server IP]
dtmf-relay rtp-nte
codec transparent
!
!
I note that if we remove "passthru content sdp" it won't connect at all.
09-24-2010 11:13 AM
No video could be due to Asymmetric Video RTP payload-type. IOS 15.1(1)T introduced Asymmetric payload type interworking :
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1403792
1) What is the IOS version used in CUBE?
2) For the 7985 video call flow, can you please provide "debug ccsip all" and "debug voip ccapi inout" with the following config changes :
voice service voip
sip
asymmetric payload full
no pass-thru content sdp
midcall-signaling passthru
Arun
09-27-2010 03:14 PM
09-27-2010 03:23 PM
Sorry - I'll just add that we are running IOS version c3845-adventerprisek9_ivs-mz.124-22.YB6
09-27-2010 03:25 PM
A number of debug outputs are missing in the file. Can you please collect the debugs in the router's logging buffer :
1) Configure the following :
service timestamps debug datetime msec
service sequence
no logging console
no logging monitor
logging buffered 5000000 debug
no logging rate-limit
2) Recreate the problem
3) Then, collect the output by entering :
term len 0
sh log
Arun
09-27-2010 03:44 PM
09-28-2010 05:03 AM
TP server sends two m=video lines in the "200 OK" response, CUBE accepts the first m=video and rejects the second one. This is normal. Snip from ACK request sent to VCS
a=ptime:20
m=video 18942 RTP/AVP 112
c=IN IP4 y.y.y.y
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=42000D
m=video 0 RTP/AVP
The TIAS parameter is not passed by CUBE in the "200 OK" SIP response sent to CUCM and in "ACK" request sent to VCS (CSCtc00502 b=TIAS parameter is not passed through CUBE for DO-DO calls). Call Signaling looks fine otherwise. Couple of things that we can do :
1) Configure the following command in the outbound dial-peer 1041 and re-try the call (work-around for TIAS issue)
voice-class sip bandwidth video tias-modifier 2304000 negotiate answer
2) Once the call is connected, check whether the 7985 is transmitting and receiving video rtp packets by pressing the "?" button twice
3) If possible, collect sniffer capture between CUBE and VCS along with the same set of CUBE debugs.
Arun
09-28-2010 03:47 PM
We are now using a different CUBE for testing purposes (a 2811) and have upgraded the IOS to 151-2.T1 as you previously reccomended.
This has resulted in some improvement - the call now stays connected as audio only, as opposed to disconnecting after around 10 seconds. Pressing the '?' button twice on the 7985 reveals that it's sending video but not recieving. I got someone to check the Telepresence server and VCS and they say they are not recieving video.
I've also implimented the voice-class config change you reccomended, but the problem is the same. I've attached new debugs. I'll try and get a packet capture soon.
Thanks very much for your help!
09-28-2010 05:39 PM
09-29-2010 07:05 AM
Thanks. With the new version, CUBE is not negotiating the video stream correctly. I will research this further. I think it will be better to track this via a Service Request and then once the analysis / findings are complete, we can post the resolution to this forum. Let me know if you would like me to create a Service request.
Arun
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