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 VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues. Today a cell phone or a POTS line phone can send DTMF clearly enough to operate a credit card or other interactive tone based system at the far end. With SIP it is sometimes "chancy". 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. Cary Fitch
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 Smith Training Manager Digium, Inc.
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 a normal POTS line. > > But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old > phones. > > Multiple transcodings cause issues. Today a cell phone or a POTS line phone > can send DTMF clearly enough to operate a credit card or other interactive > tone based system at the far end. With SIP it is sometimes "chancy". > > 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. > > Cary FitchIsn't that ulaw/alaw? j> > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
"Cary Fitch" <caryf at usawide.net> 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 identical stream through SIP/RTP or IAX. You can pick the same codecs as the PSTN uses (Alaw or ulaw, depending on country), but jitter and packet loss still makes things like DTMF or fax/modem unreliable. For DTMF it is better to signal that in RTP or SIP, for fax you want T.38, and for modems you need incense and strange rites at midnight. /Benny