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Copy of Cisco CUBE
Transcript of Copy of Cisco CUBE
What is a CUBE?
Cisco Unified Border Element
DTMF - why doesn't it work ?
A CUBE is physically the same as a a Cisco voice-enabled router.
Howerver, the CUBE has an IOS feature set with the unique capability to interconnect two VoIP call legs, as opposed to the tradditional interconnection of a VoIP call leg with a POTS call leg.
AKA - next generation IP to IP GW
The Cisco UBE is an intelligent unified communications network border element. A Cisco
UBE, formerly known as the Cisco Multiservice IP-to-IP Gateway, terminates and reoriginates both signaling (H.323 and SIP) and media streams (Real-time Transport Protocol
[RTP] and RTP Control Protocol [RTCP]) while performing border interconnection services between IP networks. Cisco UBE, in addition to other Cisco IOS Software features,
includes session border controller (SBC) functions that help enable end-to-end IP-based
transport of voice, video, and data between independent unified communications networks.
Originally, SBCs were used by service providers (SPs) to enable full billing capabilities
within VoIP networks. But the functionality to interconnect VoIP networks is becoming
more and more important for enterprise VoIP networks as well, because VoIP is becoming the new standard for any telephony solution.
-- So they added SIP suport and changed the name from IPIPGW to CUBE - My Take
Cisco UBE requires special IOS feature sets. Specifically, you need to use one of the following IOS feature sets:
INT VOICE/VIDEO, IPIPGW, TDMIP GW AES
INT VOICE/VIDEO, IPIPGW, TDMIP GW
- taken from CCVP CVoice Quck Reference 2nd Ed
Real World Example
Currently Husqvarna is using two CUBE's, each one in a different data center.
Winston Salem CUBE
8 Remote offices located throughout the US which are all connected via a Verizon MPLS cloud.
Each CUBE is connected to Verizon via MPLS as well
The Olathe CUBE provides local DIDs for Olathe and Corona CA
The Winston CUBE provides local DIDs for all the rest of the locations
Qwest also connects to the Olathe CUBE via a PIX and provides 1800 services for the call center located at that location
When using SIP DTMF needs to be configured differently.
Under the dial peers for incoming and outgoing calls:
dtmf-relay rtp-nte needs to be used which is actually RFC 2833 which encodes the DTMF key presses and other telephony events as part of the RTP (audio) stream of the SIP call.
dial-peer voice 20 voip
dtmf-relay rtp-nte (digit-drop)
its unclear if the digit-drop is needed or not - Jason A noticed that incomming calls to UCCX script would not function without it but I never tested it by removing it. Ron - i saw you did not use this, did you have DTMF issues - Unity or a UCCX system ?
Cisco states when to use digit-drop with rtp-nte
To avoid sending both in-band and out-of band tones to the outgoing leg when sending Cisco Unified Border Element calls in-band (rtp-nte) to out-of band (h245-alphanumeric).
The default signaling method when doing a SIP trunk
from CUCM is no Preference. The CUCM will pick the DTMF
method to negotiate DTMF, so an MTP is not required for the call.
if the peer endpoint supports both out-of-band and RC2833 (inband)
CUCM will negoatiate both out-of-band and RFC2833 DTMF methods. As a result,
two DTMF events would get sentt for the same DTMF keypress ( one out-of-band and the other , RFC2833)
Lets go over a few of these features
SIP Trunks to CUBEs
why use a CUBE instead of just a SIP TRUNK ?
from my perspecitve
security - CUBE masks your call manager cluster from the outside world - RTP streams and IP's
centralize transcoding from outside streams
central Gateway of all outside carriers PSTN or SIP
CUCM can only do DO and need an MTP when doing EO - most carriers want EO
What is Delayed Offer (DO) ?
- aka i want to know what codec you want to use?
What is Early Offer (EO) ?
- aka i'll tell you what codec I want to use ...
End - Questions
My Final thoughts / comments
In my experience I've only done SIP Trunk from CUCM to CUBEs
At first MTP were thought to be needed for certain services to work but
come to find out the only thing the MTP was needed for was for when an outside SIP caller (from Verizon) wanted to be transfered via consult, that was the only time an (729) MTP was noticed to be invoked.
Ron has had experience with an H323 trunk to a CUBE with CUCM 4.x and that version requires MTP for hold, transfer etc.- Ron to explain..
Call Forwarding OFFNET - Verizon only wants to see sources from its own phone numbers, so hairpining a call from an outside caller will fail. My situation I built a CFWDAll CSS that masked all calls from each location that was forwarded to the site's main number. The CUBE can do this as well and redo the SIP packet to fool Verizon and not change the caller ID but I never had time to get it to work. I have an exmaple if any body is interetested or has a chance to play with it.
SIP Rel1XX Enabled on CUCM - certain carriers will not respond with the correct SIP packet when answering..so when you call a at&t owned 1800 number say from verizon you would hear just ringining. Call Manager is expecting a 200 ok reply , instead its recieving a 183 (in progress with SDP) as an answer
.. I have CUCM screenshots..
SRST.... for sites that use SIP for internal and external calls. Need to be set up just like sites that use remote PRI's . If the network goes down then an local FXO must be at the site for 911 etc..
Global dial plan with only two gateways ? Everybody uses the same two Route Lists
If a customer wants SIP and its a new provider then they usually have to purchas a dedicated internet circuit from theat carrier so QoS can be honored - my example
Faxing... has to be all G711 in and out. Not a lot of experince with it but Jason A may can shed some light on it and T.38 with compass
Verizon does not care about 10 or 11 Digits - so how do we differenate Long Distance and local in a 10 digit local world....
SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a
conference between two or more endpoints.
Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to
an e-mail address and is in the format of sip:userID@gateway.com. The user ID can be
either a username or an E.164 address. The gateway can be either a domain (with or without a hostname) or a specific IP address.
--- taken from CCVP CVoice 3rd Ed
Debuging the SIP Call Flow on a CUBE
debug ccsip all
What is SIP ?