Cisco Jabber configuration for iOS (i phone)

Here in this post i am giving the overview of the steps that we need to do for configuring Cisco Jabber on apple iphone.

Before doing anything we need to install the cop file for jabber

1. Download the cop file from the CCO

2. Install the cop file from the menu 

3. CUCM -> OS Administration -> Software Upgrades -> Install/Upgrade

once the upload is successful reboot the server 

4. Repeat the procedure for all the subscriber 

5. Repeat the Tomcat service so that the device icon can be displayed

6. Increase the SIP Dual Mode Alert Timer value from

CM Administration -> System > Service Parameters -> [Server] Cisco CallManager (Active) -> Increase the SIP Dual Mode Alert Time to

3000 milliseconds -> Save



7. create a dedicated SIP Profile and modify the below parameter in the SIP profile to 660

    Timer Register Expires
 * Timer Keep Alive Expires
 * Timer Subscribe Expires
8. Add the Phone:

CM Administration -> Device -> Cisco Dual Mode for iPhone -> Add New ->TCT<name>

9. Add the Line to the phone and Associate the Device with the End User



10.Launch the App on the iPhone .

 

 

H323 slow start and fast start .. With Cisco IOS gateway configuration

Before discussing on this topic we must be aware of how the basic call flow occurs on h323 protocol.
here is the basic call signalling that happens with h323

here the the h225 and h245 signalling uses two different TCP connection.

In case of h323 fast start the h245 steps which is the capability and media negotiation steps occurs in the setup message of h225 step.

In normal scenario as per the above diagram can be considered as slow start as h245 negotiation is occurring with separate TCP connection .

so with fast start few message transaction occurs compare to slow start.

Now lets focus on the configuration for the same in cisco IOS gateway.

we can configure this either globally or dial peer level.

In case of global level.

1. Go to configutaion mode

2. Go to voice service mode. with command " voice service voip"

3 Configure the command "h323 call start slow" to revert to fast start configure "h323 call start fast"


In case of individual dial peer level


1. Go to configutaion mode

2. Create a Voice class using command " voice class h323 tag "

3. under the voice class command configure 'call start slow"

4. use the voice class command under the the voip dial peer  command 



Done...:-)

SIP Early offer/early Media vs Delay Offer/Delay Media

Sip is a Request/Response protocol and it uses different mechanism to establish session.This article i am going to discuss about early/delay offer used by SIP for
capability negotiation

Early offer: SIP uses SDP (Session description protocol) in order to negotiate capability between two parties.In case of the early offer the SDP message is sent from the Calling party to calllee party in the initial invite message and the callee party respond with the negotiated capability  in the 200 OK it send to the calling
 party

Delay Offer : In this case the called party send the SDP message in the 200 OK message to the calling party and the calling party respond back with the negotiated parameter in the ACK message it send to callee party

Early Media : In case of early media the media between the callee and calling party establish before the session established between the two.In normal condition the SIP endpoint play ringing sound from its local source when it recieve a ringing message from the callee party.This is sometime confusing as no one knows what is happening at the callee party,Early media play a important role here what t does is the calle party send a message 183 instead of 180 and let the calling party establish a bearer channel between the two party so that they can share the message directly.

Delay Media : In case of delay media the media is established after the SIP session negotiation is complete

CAD Login Step in UCCX

Here is what all event occurs when a CAD uses login to UCCX .

1. CAD will connect to the UCCX (LRM service) to check if there are licenses available. If it fails at this stage, we get the error "License Server is Down". The login screen is not even provided.
2. Once the user enters the userid/password, the same is sent to the UCCX. Now, the user is searched within the LDAP database of the UCCX. If the user is not present, we get the "Invalid User" error.
3. The AXL request is sent to the CUCM for authentication. Post that, the CTI Control is opened on the agent device. Here we either get the "Authentication fail" or the "Error with CTI or RMJTAPI" respentively if either fails.
4. User configuration is loaded from the LDAP of UCCX i.e. agent specific config such as integrated browser setting etc. and also the connection is made to all the services such as recording, chat, playback and so on.

SIP Message transaction

In my previous post i mentioned the basic architecture of a SIP network and the component involve in SIP .
In this post i am going to mention several messages that the SIP uses to initiate and drop a call.

SIP works on request/response model

The most common messages used by SIP are as follows

Register: UA (User Agent) send a message about its location to SIP server in register message
Invite: This message is sent by the caller in order to other agent to join a SIP call , This message can also be used in order to change any call parameter for a established call.
ACK : The SIP UA receive several response for a invite , This message acknowledge the final response
CANCEL: This message end a call that has not yet been established
BYE:  This message end a session or decline to take a call
OPTION: This message queries about the capability of the server.

Response for a request in SIP fall under the below categories:

1XX : This reponse designate a provisional and informational message
2XX: This response designate a  repsonse to a request  that is successful .
3XX : This response is used to designate a redirect or location change for the called UA
4XX: This response is used to designate a error condition from UA side (Request failure)
5XX: This response is used to designate a failure condition from the Server Side
6XX: This response is used to designate a global failure

Informational response:

100 Trying
180 Ringing
181 Call Is Being Forwarded
182 Queued
183 Session Progress

Success

200 OK

Redirection
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service

Client-error

400 Bad Request
401 Unauthorized
402 Payment Required
403 Forbidden
404 Not Found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout
410 Gone
413 Request Entity Too Large
414 Requested URL Too Large
415 Unsupported Media Type
416 Unsupported URI1 Scheme
420 Bad Extension
421 Extension Required
423 Interval Too Brief
480 Temporarily Not Available
481 Call Leg or Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable

Server-error

500 Internal Server Error
501 Not Implemented
502 Bad Gateway
503 Service Unavailable
504 Server Timeout
505 SIP Version Not Supported
513 Message Too Large

Global failure

600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable

SIP Basics

SIP is a VOIP protocol and it stands for session initiated protocol , It uses text message same as http for commnication, The SIP network consist of below component.

SIP Proxy server:The proxy server works as an intermediate device that receives SIP requests from  a client and then forwards the requests on the behalf of the client. Proxy servers can provide functions such as authentication, authorization,network access control, routing, reliable request retransmission, and security. 

Redirect Server: The redirect server provides the client with information about the next hop or hops that a message should take, and the client then contacts the next hop server or user agent server directly.

Registrar Server: The registrar server processes requests from user agent clients for registration of their current location. Redirect or proxy servers often contain registrar servers

User Agent (UA) : UA comprises a combination of user agent client (UAC) and user agent server (UAS) that initiates and receives calls. A UAC initiates a SIP request. A UAS, a server application,contacts the user when it receives a SIP request.The UAS then responds on behalf of the user. CiscoUnified Communications Manager can act as both a server and a client

SIP user uses URI in a email id format for its identification,Unencrypted SIP works on port 5060 and encrypted(TLS) sip protocol works on Port 5061

SIP Packet exchange ... Coming Next...

How Gateway discover gatekeeper

We can use gatekeeper to register different gateway to a particular zone.Each zone will have only one active gatekeeper

Now how the gateway find s its gatekeeper ???

Gateway/Gatekeeper works on h323 protocol suite and h323 protocol suite is split into 3signalling

1. h225 (RAS-Registration,admission and Status)
2. h225 control signalling 
3. h245 (Media negotaiation and logical channel or transport signalling )

H225 RAS: RAS signalling work between gateway and gatekeeper on UDP port 1719 and for multicast work on UDP port (1718). In case of multicast it used the IP 224.0.1.41 to discover the gatekeeper

H225 control signalling:This is used  to setup connection between two H323 endpoints,This signalling works on TCP port 1720

H245 Media control and transport: this signalling is used to negotiate media capabilities and parameter fpor establishing logical channel,Adminssion control is also done by h245 signalling.

Gatekeeper discovery process:

1. The gateway or h323 endpoint send a GRQ(Gatekeeper request packet ) on port 224.0.1.41 of udp port 1718
2. the gatekeeper  either respond using a GCF (Gatekeepr_ confirm) or GRJ (Gatekeeper Reject ), In case of rejection it also include the reason of rejection .