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Troubleshooting SIP in Cisco Unified communications deployments - FAQ from Live Webcast

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Introduction

deji

Ayodeji Okanlawon, a Cisco Designated VIP, is the Lead Consultant Engineer for Global Solutions Design and Engineering at Verizon Business. In his past, he has worked at Intact IS, NCS Global, and Schlumberger Information Solutions. His experience includes development of design and deployment of large scale IP telephony projects on Cisco Call Manager platforms, Cisco Voice gateways, Cisco Jabber cloud and on premise solution. His expertise includes SIP solutions, CUBE design and Deployment, Troubleshooting: Voice gateways, CUCM, Unity connection, CUPS. Deji has been awarded the Cisco Designated VIP in 2013 and 2014. Deji holds a Bachelor of Science (BS), Electrical and Electronics Engineering, Second Class Upper from Obafemi Awolowo University. 

 

You can download the slides of the presentation in PDF format here. The related Ask The Expert sessions is available here. The complete recording of this live Webcast can be accessed here.

 Cisco Ask the Expert    

There are several questions that were made during the live Webcast that are answered by Deji in the Ask the Expert Event. Make sure to check the event for more questions.

 

Understanding SIP

Q. Whats the best RFC to read for SIP?

A. RFC 3261. You can refer the url,

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7960g_
7940g/sip/7_5/english/administration/guide/ver7_5/sipaxa75.pdf

http://www.rfc-base.org/rfc-3261.html

 

Q. Is there a way to insert call id Name in the SIP header on outbound calls?

A.  Yes you can insert call id name (CNAM) in a SIP header. This is usually found in the From header.

E.g.

From: "Inspector Clouseau" <sip:011270059523@192.11.13.5>;tag=301774~23001403-1e76-4ea5-8dd4-52d6f82b946c-27747658

NB: you will need to configure the display and ascii display on the DN in CUCM for this to be present in the From header which identifies the display name the user wishes to present to other user.

However it is not enough to pass this to the ITSP, your ITSP must offer this service for it to work. The name must be registered to a valid number within their Servers for this to work.

           

Q. Do you have a recommended tool for troubleshooting SIP call flow in the gateway?

A: There are a few tools that you can use. Here are,

1. TranslatorX (http://translatorx.cisco.com/).. --You can do a whole lot with it.

2. Notepad++ (http://notepad-plus-plus.org/)        ----Helps you to analyse your logs, debugs etc

3. Agent Ransack (http://www.mythicsoft.com/agentransack) --- A powerful search tool to help you find calls within your log files (especially with CUCM logs)

 

Q.  I'm New with SIP where I can review more information?

A: Basics of SIP
http://www.cisco.com/c/en/us/tech/voice/session-initiation-protocol-sip/index.html

Basic SIP call flows 
https://supportforums.cisco.com/document/71131/basic-sip-call-flows-troubleshooting-commands

SIP Phone registration process with CUCM
https://supportforums.cisco.com/document/86036/ip-phone-sccp-sip-phone-registration-process-cucm

If you look under IP telephony on the community there are several videos, documents and material on SIP and also check out the Cisco Support Community You Tube Channel as well.

 

Q. Are there many security concern using SIP ?

A:  RFC 3261 defines ways to provide increased security for a SIP session.

The following describes areas in SIP that provides security for the protocol

1. Authenticating users.

We need to authenticate a user to ensure that the sender of the message is who he claims to be.

To achieve this SIP uses digest authentication between a UAC, proxy and a UAS. This provides the most basic level of authentication challenge between a client, proxy and a server.

2. Secure SIP signalling

The next area we can secure is SIP signalling itself. For this we use SSL/TLS. This is similar to using https in web browsers. With TLS before our any signalling is exchange X.509 certificates are

used create a secure TLS channel. All our SIP messages are then transported within the secure channel.

NB: The digest authentication mentioned above for authenticating a user agent is just authentication. The messages are not protected from reading or modification hence it is recommended that these

messages are carried inside a secure TLS channel for better security.

3. Privacy and Identification

Additional security features in SIP provides means where any user can choose to either reveal or conceal his identity.

4.Secure RTP

SIP also provides the ability to secure the media channel. It is not enough to secure signalling while anyone can listen to the media. RFC3830 discusses how the encryption should be done.

5. S/MIME

S/MIME encapsulation is used to protect sip headers making it impossible for any one in between the sender and receiver to modify the sip headers

 

Q.  Where I can find a good document on the SIP deployment model?

A:  You can refer this document for sip deployments, http://www.cisco.com/c/dam/en/us/products/collateral/unified-communications/unified-border-element/cis_45835_cube_assets_wp1e.pdf

 

Q.  In CUBE setup, sometimes I do not see reason code from remote party but CUBE add reason cause when relaying the reply to other party. Is this normal behavior?

A:  Assuming that the reason code you are referring to here is seen when a BYE or CANCEL request is sent or received.

The status code seen in the reason header field is not mandatory. It is optional. The reason header field was created to be used to share reasons why a session was terminated.

Here is an excerpt from rfc4411(https://tools.ietf.org/html/rfc4411)

"In the event that a session is terminated for a specific reason that can (or should) be shared with SIP Servers and UAs sharing dialog, the Reason Header [1] was created to be included in the BYE

Request"

You can also refer to rfc3326 for more details.

https://www.ietf.org/rfc/rfc3326.txt

This implies that Cisco has implemented CUBE to include status code in the reason header field while some other vendors have not. This is normal as it is optional to include it.

 

Q.  What is difference between P-Preferred and P-asserted Field in a INVITE message?

A:  P-Preferred-Identity:  Is used when a user has multiple Public identities.( e.g. multiple numbers)

NGN servers use the PPID header to identify the preferred number that the caller wants to use. The PPID is part of INVITE messages sent to the NGN. When the NGN receives the PPID, it authorizes

the value, generates a PAID based on the preferred number, and inserts it into the outgoing INVITE message towards the called party.

Your ITSP operates within an NGN network.

 

Q.  Is there a good debug command to determine if the carrier/etc is doing delayed vs early offer.

A:  One popular debug you can use here, "debug ccsip messages" for troubleshooting the call failure

 

Q.  I believe CUBE will be replaced by another SBC by Cisco, when is the EOL for Cube.

A:  CUBE was under some review to EOL but it's regaining some momentum with the new cloud deployments. Some functionality is being moved to Expressway but the core of CUBE will be available for quite sometime.

 

Q.  I am troubleshooting call drops. The SIP logs and SIP ladders for the dropped calls show that the SIP client is sending BYE messages before the user hangs up the phone. What would cause this? What would cause a SIP client to apparently randomly send a BYE?

A:  This could be due to SIP reinvites. Also, a session timer mismatch could cause this.

 


Related Information

 Cisco Ask the Expert    

 

Comments
Community Member

Hi ,

Thanks for such a nice information.

Links Not working :

http://www.cisco.com/c/en/us/tech/voice/session-initiation-protocol-sip/index.html

http://www.cisco.com/c/dam/en/us/products/collateral/unified-communications/unified-border-element/cis_45835_cube_assets_wp1e.pdf

Error :

The Page You Have Requested Is Not Available
http://www.cisco.com/c/en/us/tech/voice/session-initiation-protocol-sip/index.html
Were you looking for:

..

Could you please help with a document link for troubleshooting steps for various IP telephony issues.

Thanks

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