"Cary Fitch" writes:
> Is there a plain 64K codec that would simply pass through the SIP server and
> be handed off to a PRI or phone co. trunk on a T1 on the other side of the
> SIP server? Digital 64K telco sounds very good as a phone conversation.
You can't get a guaranteed bit-for-bit ident
On Thu, 12 Nov 2009, Cary Fitch wrote:
> I am not sure what the problems are and the reasons for the basic 64K modems
> used in VOIP are. I understand the compressed codecs that get the bandwidth
> down to 20-30 K. And perhaps the 64K units give much better potential audio
> than you would get
On Thu, 2009-11-12 at 08:53 -0600, Cary Fitch wrote:
> Digital 64K telco sounds very good as a phone conversation.
Digital 64k audio coming across a T1 is essentially identical to the
ulaw codec in VoIP. Digital 64k audio coming across an E1 is
essentially identical to the alaw codec.
--
Jared
I am not sure what the problems are and the reasons for the basic 64K modems
used in VOIP are. I understand the compressed codecs that get the bandwidth
down to 20-30 K. And perhaps the 64K units give much better potential audio
than you would get on a normal POTS line.
But, as phone circuits VO
