Considerations on migrating from TDM PRI's to SIP Trunking?
We are planning a migration from traditional TDM PRI's to SIP trunking for inbound toll-free and outbound LD telephone traffic. Currently, we have 7 x PRI's connected to a 3945 router feeding our CUCM 8.6 system. This works fine. We are planning on making use of a 100 Mb data circuit to bring in SIP service from our carrier. This would terminate on our 3945 voice gateway. We would have roughly the same number of SIP concurrent call paths as we have PRI channels (162).
I know that the 3945 makes heavy use of DSP resources to handle the PRI traffic. Are DSP resources needed for SIP traffic as well?
Will the SIP usage (162 SIP concurrent call paths) cause similar router utilization as the PRI's (router CPU, memory, etc.)?
What are other things we should look out for in this migration?
1. DSP resources are not required for SIP to SIP interconnectivity. DSPs are required for MTP resources if required or conference bridge/transcoder resources.
2. I would watch out for the migration period, you have 162 channels and also potentially 162 sessions. Depending on the migration strategy and the number of concurrent calls, you *might* be running the router to the limits. This also depends on what else you have running on that router. You might want to watch the memory and CPU utilization during this period.
3. You might consider upgrading IOS if you are still running older versions since there have been many enhancements on the SIP side of things.
1. Just to add to the excellent tip provided by George (+5), Here is the capacity matrix for the 3900 gateways. Your 3945 can support 950 concurrent calls. So in terms of capacity, you are well taken care of.
Number of IP-to-IP Calls per Platform
Maximum Number of Simultaneous Calls (Flow-Through)
Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
Cisco AS5350XM and AS5400XM
2. You should definitely make provisions for dsps. You may need DSPs for MTP, xcoding, etc. Especially with SIP, MTP may be needed for DTMF mistmatch, supplementary services etc
3. One of the most important thing to consider is the codec you will use for your calls. Your users are used to PSTN (TDM). Using G711 on your sip calls is not even the same as the traditional PSTN. The quality will be noticeable. Using G729 is going to be distinct. I have seen where users complain of this during a deployment. Using G729 will be a rude shock and they may not like it.
4. You also need to consider your analogue options etc FAX. What fax protocol does your ITSP support etc...We have been sued before by a customer because their fax machines didnt work. The ITSP said it supported T.38 while in reality it didnt.
5. In terms of memory utilization and CPU, I will think that it should be less for IP-IP call. This is because in an IP-TDM call, your router is constantly encoding via your dsp the IP payload (codecs) to TDM for transimissiont ot he PSTN. This wont be happening any more.
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Are you getting this error “Installer User Interface Mode Not Supported. The installer cannot run in this UI mode. To specify the interface mode, use the -i command-line option, followed by the UI mode identifier. The value UI mode identifiers...
The below trick might come handy when you have to add a new node to a cluster but you don't have or is unsure of the security password for the publisher. This procedure has been around for ages.
1) Login into the CLI of the Publisher.