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.

[abouriah10]

Dear Randy,

Could you please help me in this issue

I have Cisco E20 can not register with cisco call manager 8.6.2 and could not find the voice vlan form the switch

so could you please help

[yuanwu]

Hi,  Abouriah10

Thanks for your question.  It looks like your question was not a Cisco Unified Border Element for PSTN SIP Trunks related, instead it is a E20 registration with CUCM 8.6.2 question,  please send the question to general IP Telephony community, thanks for your understanding.

Rgds/Randy

[George Chongris]

What version of code is it running?

If its like the EX60/90's it won't pick up Voice VLAN unless you're on 5.0 code or newer

Sent from Cisco Technical Support iPad App

[Ayodeji Okanlawon]

Randy,

Its been documented that CUBE supports mid-call pass through. However in my production environment, this does not work as expected.

Here is the scenario..

fax-------CUCM-----------SIP Trunk-------------CUBE------SIP TRUNK--------ITSP

Region between Fax and Sip trunk=G711

CUBE Has inbound dial-peer from cucm and outbound dial-peer to ITSP using voice-class coded with G729 prefered.

With this scenario  fax transmission  failed. The call was setup using G729 but when fax negotitation started mid-call codec negotitation did not happen hence fax failed. AT the moment we are using prefix on RP for faxes and then created dial-peeron CUBE based on the matched prefix to use G711 for all outbound calls.

We had a discussion about this on this thread..

https://supportforums.cisco.com/message/3654743#3654743

Can you please advise us if fax transimission can work with voice-class codec, which will require a mid-call codec renegotiation.

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"

[yuanwu]

Hi, Aokanlawon

Thanks for your question.

Regarding to fax working over SIP with CUBE,  you need to decide which fax protocol mode you will use, either T38 or fax pass-through , both of them can be SIP protocol based, which means the ITSP has to support one of them, only with one of these 2 protocol configured from your fax machine under CUCM all the way to the ITSP through CUBE, otherwise you can only do G711 audio channel based fax where you achieved through your previous testing.

If ITSP can support either mode of T38 or fax pass-through ( not fax passthrough), the mid-call renegotiation can be done via CUBE to achieve your scenario, additional configuration will be needed.

If ITSP can support neither of them, then you can only do G711 audio channel based fax in which you can't upspeed the pre-established G729 codec to G711 which was required for fax transmission.

Rgds/Randy

[Ayodeji Okanlawon]

Randy,

Thanks a lot for finally clarifying this. We are using modem passthrough. The faxes are onnected using vg248 and MGCP. SO in this scenario we can only do G711 all the way?

For fax pass-through can you please help with the additional configuration or send a link to the documentation for it.

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"

[yuanwu]

Hi,  Aokanlawon

Thanks for your feedback.

Please check the following link of the different concepts for call control based pass-through, and NSE based modem passthrough.

http://www.cisco.com/en/US/partner/docs/ios/voice/fax/configuration/guide/vf_cfg_fx_passthr_ps6350_TSD_Products_Configuration_Guide_Chapter.html

There are some misunderstanding about fax pass-through, fax passthrough and modem passthrough, even at CCO since there are so many documents from time to time, some description  is not accurate.

To make a long story short, since you are using VG248, the only way you can do is , via G711u audio channel for fax,  the modem passthrough is NSE based, which is Cisco proprietary protocol, it will not be used by ITSP, the NSE packet will not be understood by the ITSP device, so you can only use G711 audio channel to make your fax work.

There will be no possibility to do Mid-call codec renegotiation for this particular setup.

Rgds/Randy

[Gordon Ross]

Are there any recommned ways to configure CUBE when having multiple different PSTN providers connecting to it ? (By default, CUBE will allow anyone to talk to anyone. So it is possible that a call from PSTN provider 1 could bounce straight back out of CUBE to PSTN provider 2 without going via CUCM.)

Or is it recommended to have separate CUBEs for each PSTN provider ?

GTG

[yuanwu]

Hi, Gordon

Thanks for your questions.

For your first question,  I think it depends on your dialplan design and you can send the call directly out to the second PSTN provider from your first PSTN provider using incoming dial-peer matching, outgoing dial-peer matching along with voice translation-profile.

For your second question, you can use CUBE HA feature to have 2 CUBEs as active and standby when you connect to multiple PSTN providers.

The above are just some general criteria,  the further design detail might needs some examples.

Rgds/Randy

[Gordon Ross]

The outbound is the trivial part. It's the inbound I'm concerned about: How do I prevent calls from PSTN Provider 1 bouncing back out through PSTN Provider 2, without first going into CUCM ?

GTG

[yuanwu]

Hi, Gordon

Thanks for your feedback.

To prevent the call from PSTN 1 to PSTN 2,  I think you can add different prefix to all the called number from PSTN1, and you can remove them via voice translation-profile while you send them to CUCM,  these modified numbers will not match the outgoing dial-peer for your PSTN2.

Rgds/Randy

[jose.albino]

Hello Randy,

I am doing some planning design for a Telephony provider regarding the best practices that can be aplied to several CUBE's with SIP Trunk integrations. And i would like to know your opinion regarding the impacto of MTP Software on a CUBE platform, namely on the maximum concurrent connections that a specific platform can handle.

Is there are any way that we can calculate this, as i know that the maximum sessions shown on a specific platform are normaly not considering that the same CUBE will implement MTP Software, for each concurrent connection.

Thanks in advance,

Regards,

José

[yuanwu]

Hi, Jose

Thanks for your questions.

Regarding to your software MTP co-location with CUBE question,  it depends on different platforms which will have different session supported, we can provide the value to you if you can name an example platform, like Cisco3925 can support around 400 connection with software MTP.

Rgds/Randy

[jose.albino]

Hello Randy,

I am interested on both the 2900 and 3900 series. I was looking for a document that would explain this details but i have not yet found one that is clarifying. Is this dependent of the RAM installed on the platform as if i increase the total RAM of the Platform would i have a higher number of concurrent connections?

For example if you take the CISCO3925 with that comes with 1GB SDRAM and with a supported Maximum of 800 SIP Session ( in a specific reference and tested scenario?), if we enable Software MTP on the CUBE that will be used for each terminated call on the CUCM, we will have a support of 400 concurrent connections?

This means that in this conditions the platform would be operating with an average CPU of around 75% of its capacity?

Regards,

Jose