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SIP Trunking
- Thread starter MitelAU
- Start date
IP networking gives you unlimited (virtually) number of cuncurrent calls limited by your network capacity. SIP requres licensing for each virtual trunk. Plus there is no route optimization when you run on SIP. Plus when you translate different signaling protocols it is always introduce some loss in functionality.
- Thread starter
- #3
Thanks Slapin
IP networking and compression licenses for small deployments (2/3 sites)which do not require clustering can be extremely expensive. 10 SIP trunk licenses and Intra-Zone compression (free)is much cheaper.
Knowledge is power, thats why we share!
IP networking and compression licenses for small deployments (2/3 sites)which do not require clustering can be extremely expensive. 10 SIP trunk licenses and Intra-Zone compression (free)is much cheaper.
Knowledge is power, thats why we share!
IrwinMFletcher
Programmer
voipin, keep in mind that the compression licenses do not go away with SIP trunks, those licenses are for the system (voicemail, eMOH, etc) and any TDM endpoints.
In other words, if you have 2 3300's connected an IP trunk, with and ONS phone on each, you'll need a compression license on each 3300 to enable G729 across that trunk when those phones are talking to each other. Changing that to a SIP trunk doesn't change the compression license requirements at all.
In other words, if you have 2 3300's connected an IP trunk, with and ONS phone on each, you'll need a compression license on each 3300 to enable G729 across that trunk when those phones are talking to each other. Changing that to a SIP trunk doesn't change the compression license requirements at all.
- Thread starter
- #5
IrwinMFletcher
Programmer
Then that's fine, my point was that you seemed to be saying that a SIP trunk would save you the cost of compression. Just pointing out that that isn't the case. If the BW isn't an issue, then you don't need it in the IP trunking case either.
IP trunking option cost is equal to 12 or so SIP trunk licenses. Plus add cost of troubleshooting SIP trunks when you will hit any sort of issues with transferring, voicemail or DTMF. SIP is relatively new for Mitel and implementation still has issues. Especially when media parameters have to be re-negotiated during existing call.
When I was playing with Policom SIP conference unit I was able to place a call into a state when original party will disconnect and 3 or more other parties will hear dead air.
I would recomend to go with a native method of interconnection which talks in the same language as every other phone or gateway and will be not translated.
When I was playing with Policom SIP conference unit I was able to place a call into a state when original party will disconnect and 3 or more other parties will hear dead air.
I would recomend to go with a native method of interconnection which talks in the same language as every other phone or gateway and will be not translated.
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