Direct SIP Trunk with CUCM

[KhammariAsma]

I have a blank problem in Communication on the SIP Trunk direct with the provider

The options selected :
* MTP required checked on trunk sip
* Early Offer support for voice and video calls (insert MTP if needed) option under SIP Profile

and i have a cisco gateway ISR that provides media resources : transcoder and mtp

the outgoing call is successful after two call attempts, on these two attempts not ok, according to the provider,there is a DATA exchange (SIG) with an @ip of my cisco gateway

What can i do in this case ? unckeck MTP required ?

[b.winter]

Honest opinion: Get rid of the direct connection between CUCM and the provider. That's nonesense.

About your problem: Probably the provider sees the IP of the GW, because it is used as the MTP endpoint, and you are forcing CUCM to use MTP.
I would create a separate MRGL / MRL for the SIP trunk, without the GW-MTP. Just the CUCM-MTP.

If you uncheck MTP required, then the media will flow directly between the provider and the endpoint (phone, jabber, ...), therefore, the provider will see your internal IP address and this can be thought of a security issue --> That's one point, why you put a CUBE between provider and CUCM / internal network.

[KhammariAsma]

Hello,

thank you for your reply.
I will try that, but the question in case the callmanager needs to do transcoding,
that will be enough for him the MTP software configured in cucm, no need to use transcoding configured in gateway,
in case of three-way conference with different codec.

Best regards,

[b.winter]

If a call needs to be transcoded, then you have the same problem as with the MTP. The GW with the Transcoder will be in the middle of the media flow.
For conferences, you could use the CUCM CFB-ressouces, then the media will come from the CUCM.

I hope you are starting to see, why a SIP trunk to the provider directly on CUCM is just none-sense. It only has disadvantages. The only advantage is, to save you the costs of an extra router and / or CUBE licenses.

[Roger Kallberg]

Possibly your mixing things up. MTP and transcoding has nothing to do with each other. Media Termination Point is one thing and Transcoding is another thing. A MTP has nothing to do with handling different codecs, that would be what a transcoder do.

[Maren Mahoney]

As has been mentioned, it is strongly recommended to have the SIP Trunk from CUCM connect to your router, and then another SIP Trunk going from your router to the service provider. This provides better call control and security for your organization.

While your router can do SIP-to-SIP call pathing without a CUBE license, having the CUBE licensed on your router will increase functionality including allowing transcoder resources to register to the CUBE function (allowing CUBE to allocate them when it determines they are necessary). 

If you can't (or don't want to) license the CUBE function on your router, you would need to work with Regions to manage transcode resources internally which can be tricky but is doable.

Let us know what we can do to help.

Maren

[KhammariAsma]

Thank you all for your feedback.
Yes I understand it is not recommended to do a direct SIP trunk with an operator, it is not a secure solution.
But, for the moment, we will keep this trunk direct.
In my case, here is the current infrastructure:
- two cucm with a direct sip trunk operator (MTP required checked, Codec G711ALAW, early offer used with insert MTP if necessary, Device pool DP_SIP having region R_SIP, the first two audio codec preference list G729 8K and G711ALAW 64K, DTMF signaling RFC 2833 )
- two cisco ISR used only for hardware media resources with cucm for transcode and conference, here is the configuration:

voice-card 0/4
dsp services dspfarm
no watchdog
!
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register XCDBRU2
associate profile 2 register CFBBRU2
!
!
no ccm-manager fax protocol cisco
ccm-manager config
!
dspfarm profile 1 transcode
description ***** Transcoder Sub1-Pub *****
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
maximum sessions 10
associate application SCCP
!
dspfarm profile 2 conference
description ***** Conference Sub1-Pub *****
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
maximum sessions 3
associate application SCCP

- in cucm, I have:
* two MTPs configured as Cisco Media Termination Point Software registered on CUCM
* two Cisco IOS Enhanced Media Termination Point transcoders for each ISR
* two Cisco IOS Enhanced Conference Bridge type conference bridges for each ISR

- in TRUNK sip direct, I have an MRGL_SIP which contains MRG MTP from cucm and MRG transcoder and MRG conference from ISR

If I'm not mistaken, with this direct sip trunk, if the cucm needs to transcode or make coferences with three or more, this MRGL_SIP must be associated with the sip trunk, otherwise it won't work?

The problem that on three consecutive calls, two calls do not work and there is a gap in the communication, on these calls not ok, the provider received data exchanges (sig) on one of the ISR IPs, perhaps because the 'MTP required option is checked.

Can this problem be solved if I uncheck the MTP required option? or should we ask the operator to add missing routes with these ISRs?
In any case, I need the transcoder and conference resources in my infrastructure.

What do you think ?

Thanks in advance

[Roger Kallberg]

Why do you not use your two ISRs as SBCs? It’s pretty much a lost cause to try to get your current setup to work reliably.

[KhammariAsma]

These are 3800 series ISRs, do not support SIP trunk, in the long term we will replace them with SBCs and configure SIP trunk in these SBCs.
For the moment, we are using a direct SIP trunk and I would like to resolve the problem of blanks in the communication