Ask the Expert:Cisco Unified Border Element for PSTN SIP Trunks

[ciscomoderator]

With Randy Wu

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn from Cisco expert Randy Wu  best practices on how to configure and troubleshoot Cisco UBE for the public switched telephone network Session Initiation Protocol trunks.

Randy Wu is a senior customer support engineer in the Multiservice Voice team at Cisco in Sydney. He has vast experience and knowledge configuring, troubleshooting, and designing Cisco UBE, gateways, and gatekeepers, working with H323, MGCP, and SIP protocols. He joined Cisco as a systems engineer in 1999. He holds CCIE certification (#8550) in Service Provider, Routing, and Switching and Voice.

Remember to use the rating system to let Randy know if you have received an adequate response. 

Randy might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the  Collaboration, Voice and Video sub-community discussion forum shortly after the event. This event lasts through June 29, 2012. Visit this forum often to view responses to your questions and the questions of other community members.

[Nizar Abuseni]

Hello Randy,

I just tried your workaround but with no luck.

Thanks,

Nizar

[yuanwu]

Hi, Nizar

Please attach "show tech" of the CUBE when you applied the workaround command.

Rgds/Randy

[Nizar Abuseni]

Hello Randy,

Today I did the tests again, and seems everything is working fine. The two commands of workaround fixed the issue, also when I did the upgrade the CUBE is more stable for handling calls.

I have tried the workaround in an old version of IOS and in a new version of IOS and in both cases its working, but in an old version sometimes its not detecting the INVITE but in a new IOS everything is stable.

But my question is why the INVITE is fragmented? is it something Huawei should fix? or it is normal because of UDP transport?

Thanks alot.

Regards,

Nizar

[yuanwu]

Hi, Nizar

Nice to hear the issue was solved.

1. the INVITE message was fragmented due to the frame is bigger than the Ethernet's MTU

2. Actually it is ok to send this kind of big INVITE packet, from RFC 3261 it can be maximum 65,536 bytes.

3. Some versions of the Cisco IOS got the issue to reassembly this kind of  fragmented packet, that is why you need to use the workaround or upgrade the IOS.

Rgds/Randy

[Ayodeji Okanlawon]

Nizar,

Just to ask how are you trying to collect or see the logs. I noticed you have loggin to buffer enbaled but only warning messages are configured to loggin.

Can you try this

no logging buffered 51200 warnings

logging buffered 10000000 debug
no logging console
no logging monitor

Then do a test with your debugs on. After you complete the test..

type.........sh loggin  (and see if you see any debug in your buffer)

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

[Nizar Abuseni]

Hello,

Actualy I enable the debugs and writing the logs into text file using Putty. I can see the debugs for calls from CUCM to CUBE without any problem. The problem was from SIP provider, they were sending the INVITE message as fragemented messages.

Thanks for your suggestion.

Regards,

Nizar

[ksubrama]

What is the recommended way of implementing high availability with CUBE

in terms of redundant CUBE elements for example?

[yuanwu]

Hi, Ksubrama

Thanks for your question.

Regarding to the redundancy of CUBE,

      You can implement a wide variety of redundancy designs of varying levels of sophistication using combination of capabilities offered by Cisco Unified Border Element. The redundancy building blocks include inbox software redundancy (Cisco ASR 1001, 1002 and 1004 platforms), inbox hardware redundancy (Cisco ASR 1006 platform), box-to-box redundancy (Cisco ISR G2 and ASR 1001, 1002 and 1004 platforms. The box-to-box redundancy features on the ISR G2 platforms preserve established calls during a failover, the inbox and box-to-box redundancy features on the ASR platforms provide stateful failover. Deploying box-to-box redundancy on the ISR G2s requires a minimum Cisco IOS release of 15.1.2T, and on the ASR platforms a minimum of Cisco IOS-XE Release 3.2.

Please refer to the following URL for CUBE HA configuration for ISR G2.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080b40d82.shtml

Rgds/Randy

[ksubrama]

Hi  Randy,

Thanks for the detailed reply. I am assuming box-box redundancy supports active-active modes.

For box to box redundancy do you see customers using load balancer

in case they do not use HSRP as described by the solution note you pointed to?

regards,

Krishnan

[yuanwu]

Hi, Krishnan

Thanks for your feedback.

1. box-box redundancy is active-standby mode, only the active box will process the calls

2. For load balancer, you can use Cisco Unified SIP Proxy (CUSP) to equally distribute the calls.

Rgds/Randy

[micromedicreturns]

Dear Randy,

Thank you for spending your time on this event!

Please  advise on this situation, as I have not previously implemented a router  as a CUBE.  We had a Cisco UC560 running skinny phones, and connecting  to an ITSP via a SIP trunk on a public interface.  The local IT team  on-site then installed a Asterisk/Trixbox server on the LAN that had  some SIP phones attached to it, and added a dial-peer on the UC560 to  enable the phones to call each other.  The situation is that calls can  be made between the two phone systems, and incoming calls from the ITSP  can be completed to SIP phones on the Trixbox.

What  does not function is outgoing calls to the ITSP from the SIP phones  attached to the Trixbox.  The TB is treating the UC560 as a SIP gateway,  and in fact the call does actually set up, but with no audio in either  direction.  A debug of the rtp packets on the UC560 reveals that during  the call, the CUBE is receiving packets from both the ITSP and the TB on  the LAN, but is not transmitting any packets in either direction.

Can  I ask if it is possible to setup the UC560 in this role, given that it  already has one SIP trunk configured? I had heard that a limit of one  "trunk" per router is built-in(?).  If it is possible, should be perhaps  configure it as an H.323 gateway, to provide the distinction between  the two IP systems?

[yuanwu]

Hi, Micromedicreturns

Thanks for your questions.

1. since UC560 is not an officially supported platform for CUBE function, it will be only best effort for related function.

2. From technical point of view, if I understood correctly,  you got no way audio if SIP phones under Trixbox makes outgoing call to the ITSP while there is no issue when ITSP caller makes incoming calls to the SIP phones under Trixbox via UC560?  If the configruation is symmetric, then it is quite interesting for this asymmetric symptom.

3. For supported platform for CUBE, there was no such one trunk per router limitation.

Rgds/Randy

[micromedicreturns]

Hi Randy,

Thank you for your response.  We did suspect that it was perhaps not fully supported, but as the commands were available we thought it might be...

As far as I can see the configuration is symmetric, and the UC560 appears to also transcode successfully for the incoming calls.  It is a little odd.

Regarding the trunk limitation, I think I was getting confused with the sip-ua configuration, in that only one set of credentials could be configured.  You are right  - you can direct SIP to various IP addresses from one router.

Thanks again for your help,

Best regards,

MM

[yuanwu]

Hi, Micromedicreturns

Thanks for your feedback.

Regarding to your credential command under sip-ua confusion, please check the following feature which might help you,

Configuring Multiple Registrars on SIP Trunks

http://www.cisco.com/en/US/partner/docs/ios-xml/ios/voice/sip/configuration/15-1mt/Configuring_Multiple_Registrars_on_SIP_Trunks.html#GUID-BB6EE70E-1410-453A-A65E-91505345C8EB

Rgds/Randy

[micromedicreturns]

Hi Randy,

Great, thank you for the advice and for the link - I will read up on this further.

All the best,

MM