Re: [TIProtocol-askcisco] Teliris & TIP Protocol Start-up Issues

SourceForge Headquarters 1320 Columbia Street Suite 310 San Diego, CA 92101 +1 (858) 422-6466

Aravind

Two issues popped up in our quick review ...

1)   Not enough bandwidth in the offer for Video; 320K. 

Venturing a guess, have you configured the CUCM's default region
bandwidth?  

If not, do the following after logging in to the CUCM

   System-->Region, click "find", then the default region will show.
Click "default".
   Go to "Modify Relationship to other Regions, and select  "Default".
   Under Max Video Call Bit Rate, enter "32000", and  select the "kpbs"
radio button.
   Click "Save", then "Reset".

2)   The ACK SIP to our SIP 200 OK only indicates G.711, so be sure you
offer/ack AAC-LD first or only

m=audio 53008 RTP/AVP 0^M
b=TIAS:64000^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M

From: Aravind Sethuraman [mailto:ara...@te...] 
Sent: Tuesday, April 19, 2011 2:01 PM
To: David Benham (dbenham)
Cc: ste...@te...; tip...@so...
Subject: Re: Teliris & TIP Protocol Start-up Issues
Importance: High

David,

Thanks for getting back..
As per your request attached are the invite.txt and the cts log files
this time, the behaviour was CTS reports Configuration mismatch....

much obliged
Aravind

On 04/19/2011 04:12 PM, David Benham (dbenham) wrote: 

Aravind,

Send us a full copy of your initial SIP INVITE containing the SDP. 

Also send a copy of the CTS' logs just in case.

From: Aravind Sethuraman [mailto:ara...@te...] 
Sent: Tuesday, April 19, 2011 9:31 AM
To: ste...@te...
Cc: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement

Hello David
Firstly i would like to thank you for offering your help. 
What we are building is a TIP user agent plugin which acts as a B2BUA
talking TIP on one side and SIP/H264SVC on the other side. Our ultimate
goal is to build this to interop with Cisco CTS family of products,
1000-3000. We are currently testing with a CUCM version 8.5 and CTS 1000
version 1.7 and we have chosen to adopt TIP profile version 6.

The TIPUA does support the following

1.	SIP signaling 
2.	Video

	1.	 supports H264
	2.	supports b=TIAS 
	3.	supports RTP format 112 and packetization mode
	4.	supports the profile-id - 4d0028 

3.	Audio

	1.	AAC-LD (we are faking it currently but @ signal level we
are doing it to make CTS happy. We will eventually have this up and
running)
	2.	G711

4.	KeyPress

	1.	RFC2833 ONLY

5.	TIP Negotiation -> We are having the TIP negotiation module
which i will define further below...

What we are seeing is as follows:-

1.	Our TIPUA sends invite to CTS (both are registered to CUCM) 
2.	CTS sends its SDP in 183 
3.	upon receipt of 183, we start the TIP negotiation -> note, We
are assuming the RTCP port available is RTP+1 (in the SDP)

	1.	We are using the libTIP version 1.3 library. 

4.	eventually CTS sends a mid-call re-invite and downgrades to
audio only call.
5.	We have not been able to capture ANY RTCP/TIP packets from the
CTS leading to eventual failure of TIP and therefore AUDIO only call.

I am at my wits' end. Are we doing anything glaringly wrong? does CTS
1.7 NOT support TIP ?(highly unlikely)

I am also available for any phone call if you feel it is necessary to
have @ 1917-355-0119

much obliged
Aravind Sethuraman

On 04/13/2011 05:42 PM, Steven Gage wrote: 

David,

Aravind is our Chief Software Architect and the technical lead on our
Interop efforts. 

Aravind,

David has been extremely helpful in providing direction on TIP
Interoperability. 

Please document our assumptions specifically re sip and tip end point
connectivity, minimum bandwidth requirements for video, and audio codec
requirements. 

At David's suggestion we will be attending superop but would like to get
the basics working in our lab. 

Regards,

Steven Gage
Teliris
55 Broadway | NY NY 10006 
O+1.212.490.1065 x1400
F+1.212.202.5432 
M+1.917.952.2212

------------------------------------------------------------------------
----

This message is a PRIVATE communication. This message and all
attachments are a private communication are confidential or
protected by privilege. If you are not the intended recipient, you are
hereby notified that any disclosure, copying, distribution or use of the
information contained in or attached to this message is strictly
prohibited. Please notify the sender of the delivery error by replying
to this message, and then delete it from your system. Thank you.

________________________________

From: "David Benham (dbenham)" <db...@ci...>
<mailto:db...@ci...>  

Date: Wed, 13 Apr 2011 12:58:02 -0700

To: Steven Gage<ste...@te...>
<mailto:ste...@te...> 

Subject: RE: Teliris & TIP Protocol License Agreement

Hi Steven

The TIP protocol specification does recommend ("SHOULD"), but does not
mandate, that endpoints default back to a p2p video and audio call if
TIP is not negotiated.  The CTS 1000 you have will default back to an
audio only (G.7xx) call, in such a case.  

Also, when you do attempt to negotiate TIP with that CTS 1000, you need
to indicate AAC-LD in the SDP audio line (per sect 3.1 in the Cisco TIP
Endpoint Implementation Profile).  Otherwise, the CTS endpoint will
assume it is not a TIP session and resort to an audio-only call.

Let us know if this helps or can answer more questions.

From: Steven Gage [mailto:ste...@te...] 
Sent: Wednesday, April 06, 2011 6:56 PM
To: David Benham (dbenham)
Subject: Re: Teliris & TIP Protocol License Agreement
Importance: High

David,

I posted our issue, here is a synopsis:

We are trying to setup call between a CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42).

The Mirial client is a pure SIP Video Endpoint capable of supporting
H264 AVC and G711.

We have registered it to the CUCM (version 8.5) as a 3rd party Advanced
SIP Endpoint with Digest authentication

We have a CTS 1000 registered to same CUCM

While trying to setup call between the CTS 1000 and SIP Video Endpoint
(Mirial ver 7.0.42), we are not able to setup a video call. We see in
the SIP Packet Transfer, that CUCM re-writes (there are 2 

SIP Invites sent, but that gets fairly in depth SIP wise) the invite as
audio only invite.  

Basically we need to figure out what is the configuration (do we have to
add a SIP Trunk etc) on the CUCM to make it recognize/support video
calls with a pure SIP Video Endpoint.

Thanks,

Steve

On Apr 2, 2011, at 10:36 AM, David Benham (dbenham) wrote:

Steven

Considering it is only 6 weeks away, I strongly suggest attending IMTC's
SuperOp event, where Cisco and Polycom (at least) will have gear set up
to test with each other's TIP products.    Non-members are invited to
this SuperOp, but do note that if you join IMTC, you can also partake in
the TIP Activity Group's stewardship of TIP as well as other membership
benefits.

Here are links and email addresses for further query.  

IMTC page for TIP Developers

http://www.imtc.org/tip/

IMTC page for TIP Activity Group

http://www.imtc.org/activity_groups/tip.asp

Places to post questions   tip...@im...

   or at the TIP open source project

http://sourceforge.net/projects/tiprotocol/support

IMTC page for SuperOp event (top item on page)

http://www.imtc.org/events/

Cisco page for TIP Developers

http://www.cisco.com/go/tip