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New Member

CUCM 6.1 and incoming SIP calls

Hi!

I have a task to forward incoming calls from a distant region to our central office. Our voip provider in the distant region collects calls from PSTN and then forwards them to the central office via ip-vpn by SIP (g729 codec), where we have a gateway (cs2811/pvdm) and CUCM cluster. So the thing is about only about INCOMING sip calls.

And the question is which is the better way to process those calls? To collect them at the gateway and then forward to CUCM or to setup a SIP trunk at CUCM and process them right there, bypassing the gateway?

Thnx!

1 ACCEPTED SOLUTION

Accepted Solutions

Re: CUCM 6.1 and incoming SIP calls

This is another way of asking "Should I put a CUBE between my CUCM and my SIP trunk or not?".

This was address in Christina's SIP trunk Ask-the-Expert session that I believe is still active.

Stolen from her post, these are the reasons:

" Reasons to terminate a SIP trunk on an enterprise demarc such as CUBE include but are not limited to:

- Lack of call admission control (SLA enforcement and DOS attack mitigation) on the SIP trunk

- Visibility of the CUCM and endpoint IP addresses to the SP network (and therefore to potential hackers)

- Very limited SIP trunk load balancing and redundancy capabilities

- No SIP trunk sharing between multiple CUCM clusters or other IP-PBX/proxy call agents in the enterprise

- No SIP malformed packet or other protocol level attack mitigation for your CUCM

- No way to troubleshoot voice quality problems to determine if it's your network or the SPs network at fault

- Much more limited toll fraud prevention techniques on the SIP trunk

- No way to control IP QoS settings on the incoming packets from the SP, and no way to customize them on the outgoing packets

- No way to manipulate SIP msging from the SP before it hits your CUCM to customize it to what CUCM/IP-PBX prefers to see

- Limited means of complying to the SP UNI (SIP msg manipulation on outbound msgs to the SP, and capabilities such as early-offer)

- Having to implement the SP UNI on CUCM instead of your enterprise preferred policies (and having to replicate this on every CUCM and IP-PBX routing calls to the SIP trunk)

- Having no way of doing a SIP registration to the SP when this is required on the SIP trunk "

-nick

2 REPLIES

Re: CUCM 6.1 and incoming SIP calls

This is another way of asking "Should I put a CUBE between my CUCM and my SIP trunk or not?".

This was address in Christina's SIP trunk Ask-the-Expert session that I believe is still active.

Stolen from her post, these are the reasons:

" Reasons to terminate a SIP trunk on an enterprise demarc such as CUBE include but are not limited to:

- Lack of call admission control (SLA enforcement and DOS attack mitigation) on the SIP trunk

- Visibility of the CUCM and endpoint IP addresses to the SP network (and therefore to potential hackers)

- Very limited SIP trunk load balancing and redundancy capabilities

- No SIP trunk sharing between multiple CUCM clusters or other IP-PBX/proxy call agents in the enterprise

- No SIP malformed packet or other protocol level attack mitigation for your CUCM

- No way to troubleshoot voice quality problems to determine if it's your network or the SPs network at fault

- Much more limited toll fraud prevention techniques on the SIP trunk

- No way to control IP QoS settings on the incoming packets from the SP, and no way to customize them on the outgoing packets

- No way to manipulate SIP msging from the SP before it hits your CUCM to customize it to what CUCM/IP-PBX prefers to see

- Limited means of complying to the SP UNI (SIP msg manipulation on outbound msgs to the SP, and capabilities such as early-offer)

- Having to implement the SP UNI on CUCM instead of your enterprise preferred policies (and having to replicate this on every CUCM and IP-PBX routing calls to the SIP trunk)

- Having no way of doing a SIP registration to the SP when this is required on the SIP trunk "

-nick

New Member

Re: CUCM 6.1 and incoming SIP calls

Thanks for rapid answer!

Really a lot of reasons not to have a direct SIP trunk.

That means I'll have to spend a little time configuring transcoding feature, because now there seems to be a problem with g729<->g711 codecs. The phone rings, but the connection is being dropped after several seconds.

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