One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his notebook and Xlite, but also tried the snom 360. Any hints? He also told me that he used another sip service before with the same bad result. I wonder if the Kaza boys have here something built in, .... bye Ronald Wiplinger
> One of my user is praising Skype!!! > > I cannot figure out anymore what I can improve! > > This users sip show peers is jumping from 65 msec to 1800 all the time. > Of course his voice quality is like a morse code with dashes or dots of > connection time. > The next minute he calls me via Skype and it works fine !!!! What > indicates that there is no fault on his Internet connection!!! > > He is using his notebook and Xlite, but also tried the snom 360. > > Any hints?Is he calling you on another VoIP phone or calling you on a landline/cellphone (through the PSTN)? If he is calling a landline/cellphone, then it is probably your upstream termination provider that is having jitter problems (this is my exact problem). If I check my voicemails on my IP phone (which connects directly to my asterisk box 60 miles away), everything is great. HOWEVER, if I *dial* my telephone number and check my voicemails (as if I was calling in to check my voicemails), I get loads of jitter. So between my IP phone and my * box, the connection is great, but its what is after my * box that is causing the problem. Who is providing you termination? - Gabe
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote:> One of my user is praising Skype!!! > > I cannot figure out anymore what I can improve! > > This users sip show peers is jumping from 65 msec to 1800 all the time. > Of course his voice quality is like a morse code with dashes or dots of > connection time. > The next minute he calls me via Skype and it works fine !!!! What > indicates that there is no fault on his Internet connection!!! > > He is using his notebook and Xlite, but also tried the snom 360.Skype uses iLBC codec, which has great jitter compensation. IIRC, the newer SIP channels of * are supposed to have the same capabilities, but I have not tested. I really do not like Skype (prefer FWD), but I must say, over satellite, etc, they provide quality.. All about the codec in this case.. -Greg
Ronald Wiplinger wrote:> One of my user is praising Skype!!! > > I cannot figure out anymore what I can improve! > > This users sip show peers is jumping from 65 msec to 1800 all the time. > Of course his voice quality is like a morse code with dashes or dots of > connection time.If that's what is going on (and other users don't have a problem), then it's likely to be a connectivity issue somewhere between the user's ISP and your own. A traceroute to his IP address may help you to identify the issue.> He also told me that he used another sip service before with the same > bad result. I wonder if the Kaza boys have here something built in, ....Perhaps the other sip service used the same ISP that you do... or his is somewhat flaky but happens to have a good connection to Skype... We keep our equipment in a very well-connected data center, because problems like that will kill your service and there's not much you can do to fix it (besides move). Yours, Yaakov Menken -- Yaakov Menken Capalon Internet Solutions Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell menken@capalon.com
> -------- Original Message -------- > > Skype uses iLBC codec, which has great jitter compensation. IIRC, the > newer SIP channels of * are supposed to have the same capabilities, but > I have not tested. I really do not like Skype (prefer FWD), but I must > say, over satellite, etc, they provide quality.. All about the codec in > this case..Errr...no...this is wrong. Skype uses ISAC from Global IP Sound. iLBC is something different see http://www.globalipsound.com/solutions/solutions_Codecs.php One of the reasons Skype sounds good is that its a closed system and so can leverage a wideband codec. Instead of the normal 8khz sample rate it uses 16khz. That makes for clearer sound. Since ISAC is a proprietary relative of iLBC its jitter compensation is also very good. My understanding is that Asterisk cannot presently use any wideband codecs as it is hard coded to the 8khz sample rate at its core. Adapting Asterisk to wideband capability has been discussed but will be a huge amount of work. Further, only if you know that the calls will stay wideband end-to-end will the benefits of wideband be apparent. That means no PSTN segments. Michael Graves mgraves@mstvp.com
mgraves@mstvp.com wrote:>> -------- Original Message -------- >> >> Skype uses iLBC codec, which has great jitter compensation. IIRC, the >> newer SIP channels of * are supposed to have the same capabilities, but >> I have not tested. I really do not like Skype (prefer FWD), but I must >> say, over satellite, etc, they provide quality.. All about the codec in >> this case.. >> > > > Errr...no...this is wrong. > > Skype uses ISAC from Global IP Sound. iLBC is something different see > http://www.globalipsound.com/solutions/solutions_Codecs.php > > One of the reasons Skype sounds good is that its a closed system and so > can leverage a wideband codec. Instead of the normal 8khz sample rate > it uses 16khz. That makes for clearer sound. Since ISAC is a > proprietary relative of iLBC its jitter compensation is also very good. > > My understanding is that Asterisk cannot presently use any wideband > codecs as it is hard coded to the 8khz sample rate at its core. > Adapting Asterisk to wideband capability has been discussed but will be > a huge amount of work. Further, only if you know that the calls will > stay wideband end-to-end will the benefits of wideband be apparent. > That means no PSTN segments. > > Michael Graves > mgraves@mstvp.com > >Sadly to say, but users do not care about the why, they only care about the quality! and they simple ask to "fix" it! I hope there is soon a solution, otherwise, we have to skip all our effort and just use skype!!!!! And I would hate to see that. I just lost 20 US$ to Ebay - the newly parent company of skype, for a not received parcel, but the rules says, below 25 US$ there is no guarantee that you get anything!!!! bye Ronald Wiplinger
What would be ideal is the introduction of an open source wideband codec implementation. Then you could see it adopted into SIP end points and used with SER realtively quickly. Sadly, an Asterisk implmentation would lag a little behind due to the amount of work required in an implementation that processed the streams to bridge into the TDM/PSTN world. It would be great....but don't hold your breath. For now there are Skype bridges like PSWG and Uplink that interface Skype to SIP. These are simplistic but sometimes workable. Does anyone here have experience with Uplink? I tried PSGW and gave up eventually. Michael Graves Sr Product Specialist Pixel Power Inc mgraves@pixelpower.com o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262> -------- Original Message -------- > Subject: Re: [Asterisk-Users] Compare to Skype > From: Ronald Wiplinger <ronald@elmit.com> > Date: Sun, April 30, 2006 9:09 am > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > > mgraves@mstvp.com wrote: > >> -------- Original Message -------- > >> > >> Skype uses iLBC codec, which has great jitter compensation. IIRC, the > >> newer SIP channels of * are supposed to have the same capabilities, but > >> I have not tested. I really do not like Skype (prefer FWD), but I must > >> say, over satellite, etc, they provide quality.. All about the codec in > >> this case.. > >> > > > > > > Errr...no...this is wrong. > > > > Skype uses ISAC from Global IP Sound. iLBC is something different see > > http://www.globalipsound.com/solutions/solutions_Codecs.php > > > > One of the reasons Skype sounds good is that its a closed system and so > > can leverage a wideband codec. Instead of the normal 8khz sample rate > > it uses 16khz. That makes for clearer sound. Since ISAC is a > > proprietary relative of iLBC its jitter compensation is also very good. > > > > My understanding is that Asterisk cannot presently use any wideband > > codecs as it is hard coded to the 8khz sample rate at its core. > > Adapting Asterisk to wideband capability has been discussed but will be > > a huge amount of work. Further, only if you know that the calls will > > stay wideband end-to-end will the benefits of wideband be apparent. > > That means no PSTN segments. > > > > Michael Graves > > mgraves@mstvp.com > > > > > > Sadly to say, but users do not care about the why, they only care about > the quality! and they simple ask to "fix" it! > > I hope there is soon a solution, otherwise, we have to skip all our > effort and just use skype!!!!! > And I would hate to see that. I just lost 20 US$ to Ebay - the newly > parent company of skype, for a not received parcel, but the rules says, > below 25 US$ there is no guarantee that you get anything!!!! > > > bye > > Ronald Wiplinger > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users