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ciscomoderator
Community Manager

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

Read the bioWith 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.

85 REPLIES 85

Hi Randy,

We did switch to H.323 on our Trixbox to UC560 configuration, and now the calls are functioing in both directions. 

Thanks again for the advice,

MM

hi, MM

Thanks for your update about solving the issue.

Rgds/Randy

Dan Rogers
Beginner

Hi Randy,

My question is related to the sizing of CUBE(s) relative to their function. Many of my customers wish to use CUBE routers not only to terminate their SIP Trunks, but also as WAN/MPLS routers, put VPN encryption on the device and/or use it as a L3 routing device.

In specific, what is a best practice design around this? For SMB customers I can see the value here, but for larger customers that may have Gig WAN links, I would assume that the overall throughput of the router would be the biggest determining factor, in which case an ASR would be required. Is there any documentation I can take to my customers saying "if you are going to do encryption/WAN/additional services on your CUBE this is the impact and this is the sizing tool I used to come up with the recommended ISR/ASR for all of this.

Also, for PRIs DSP allocation is easy. For SIP trunks I believe it comes down to system resource availability and whether or not hardware/software MTP is required. Are there any general rules of thumb to be aware of when allocating DSPs for CUBEs? I.E. If ITSP requires hardware MTP termination and your BHCA is X, then use this value - (or is the DSP calculator still the only tool to use)?

Hi, Dan

Thanks for your questions.

1.  For the request to enhance the performance of the router when combine CUBE feature with traditional  WAN/MPLS router plus other VPN encryption at the same router , our development team is working on the project.

2. For the large customer,  currently we recommend to seperate the funtion between CUBE and WAN router.  The sizing tool will be available when you consult your local System Engineer for the project.

3.  Again, the sizing tool with other features enabled at the same CUBE router like  MTP resource, VXML session termination will be available when you contact with your local System Engineer.  It related to company confidential policy, I can't reveal them here.

Thanks for your understanding.

Rgds/Randy

Nizar Abuseni
Cisco Employee

Hello Randy,

I am facing a strange issue in our SIP trunk. The problem is that sometime (randomly)  when I call from IP Phone to PSTN through the SIP trunk it fails, it happends randomly, some of the calls are working fine.

For example if IP Phone calls 907700368064 calls succeeds. Again if the same IP Phone calls same number it will fail. I have tried this from many IP phones and the issue was same.

I have attached sh run, debug of working call (ccsip all and voice ccapi inout) debugs.

What I see the difference between the two calls in debugs is that in non working call in application/sdp I see (a=inactive) while I dont see this in working call debug.

Also I am not sure if this is related or not, but when I do incoming call from SIP to CUBE, I see error:

Jun 21 18:50:46.023: %SIP-3-UNSUPPORTED: Unsupported ptime value
Jun 21 18:50:46.023: //482/D8427136823E/SIP/Info/sipSPIDoPtimeNegotiation: Unsupported Ptime value 5 :: using minimum ptime value of 20


I am not sure if this could affect outgoing calls, but for incoming everything is working even with the ptime error.

The call flow:

IP Phone -- CUCM -- SIP Trunk--  CUBE --SIP Trunk -- SIP

Codec from IP Phone to CUCM SIP Trunk (G729) , there is transcoders associated with SIP Trunk and IP Phones also.

Codec from CUCM SIP Trunk to CUBE is G711ulaw.

Many thanks,

Nizar

Hi, Nizar

Please provide the following information,

1. show tech of the CUBE,   10.200.128.133 is the ip address for transcoder?  please provide related component's ip address.

2. "debug voip ccapi inout", "debug ccsip message" in one file for good, bad call

3.  "debug voip ccapi inout", "debug ccsip all" in one file for good, bad call

4.  the "a=inacive" is an issue, need to check CUCM trace to see why CUCM sending "a=inactive"

5. What is the ip phone model?  SIP, SCCP?

6.  the ptime error seems just warning,  the minimum ptime for g729 codec is 20ms in IOS.

Rgds/Randy

Hello Randy,

Kindly find attached file, it includes all requested logs from CUBE and CUCM.

Also I have attached one of the transcoder's configuration, we have 3 voice gateways that are used from transcoders and they all have identical configuration for transcoder.

IP Address of transcoders: 10.200.128.132, 10.200.128.133, 10.200.128.134

CUCM IP Address that IP Phones (CIPC, 7940) are registered 10.200.128.120

CUBE IP Address: 10.200.128.130

Many thanks,

Nizar

Hi, Nizar

after reviewing the debugs and CUCM trace,

1.  The a=inactive issue was caused by CUCM,  it might be related to the transcoder being used as the MTP resource, please open a CUCM Sip issue TAC ticket to find the root cause, for the workaround, you can uncheck MTP required option in Sip trunk configuration in CUCM since the end point can use g.729 all the way to the ITSP via CUBE,right?

2. The other thing you can try,   Since CUCM sends a=inactive in SDP, the CUBE just passed on it to the ITSP, where the ITSP replied with 183 message in which the rtp port is 0, the CUBE released the call due to this rtp port 0,  you can use sip profile in CUBE to change a=inactive, or remove this line, try to see if the behavior from ITSP will be different.

v=0

o=HuaweiSoftx3000 1073741826 1073741827 IN IP4 192.168.155.148

s=SipCall

c=IN IP4 10.172.172.1

t=0 0

m=audio 0 RTP/AVP 0 101     <------  RTP port should not be 0

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=inactive

Rgds/Randy

israel.roybal
Beginner

Hi Randy,

Do you know if they are planning on adding an SCB interface command to ISR routers, I have used it on ASR and it is really useful.

Hi,  Israel

Thanks for your question.

I wonder if you refer to SBC interface in ASR platform you mentioned.

If it is the case,  since ASR 1K is a different infrastructure platform for CUBE function, even most of the command used for enterprise version of ASR is same as the ISR CUBE, the architeture is complete different.  The SBC interface will be only available at ASR platform, will not be available in ISR platform.

Rgds/Randy

mynewlogin
Beginner

Dear Randy,

Could you please tell me which DTMF methodes are supported by Cisco CUBE?

Can the CUBE functionality use local DSP resources or DSP resoursec from another Cisco Routers for DTMF issues solving?

Andrii

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