[asterisk-biz] SIP PSTN Gateway Viable?

[kenny.kant at running-config.com]

Hello All,

We have been doing Asterisk and CME implementations recently but we  
almost always exlusively bring in analog lines and or PRI for PSTN  
access to our systems.  I have known about providers providing SIP  
based lines and SIP trunks to end users for PSTN access.  I am curious  
about the following:

- How practical is this?  The idea of terminating pstn calls to across  
the Internet which is an unguarenteed medium concerns me.  Even if our  
access to it is quazi stable T1 data type of access.  Do any of you do  
systems where this is soley the method used for incoming calls from  
the pstn?  If this is done are there things to look for in a SIP  
provider, as in their presence on the Internet latency ..etc?

- What are the major advantages?  I know some places provide all you  
can eat plans which could be seen as a plus and some others provide  
really low rates. Are there others?

- Who are the major players?  How are these usually ordered and identified?

- Any general tips?

Thanks all!