Re: [Asterisk-Users] Compare to Skype
On Apr 30, 2006, at 9:03 AM, Eric ManxPower Wieling wrote: There are 2 issues here. 1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to transport audio for SIP (and other protocols). This means that ANY jitter on the SIP Phone - Asterisk link will cause audio problems. 2) Asterisk times it's outgoing audio based on the incoming audio. Therefore, if there is jitter on the SIP Phone - Asterisk link then Asterisk will replicate that jitter on the Asterisk - SIP Phone direction. In my experience, even if you have two asterisk systems with the async timing patch applied and are using IAX with the jitter buffer enabled, asterisk STILL cannot compensate for jittery links as well as Skype can. I take this to mean that the asterisk jitter buffer needs more work. In addition to having a better jitter buffer, Skype also clearly has wideband codecs which sounds better. ___ --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
Re: [Asterisk-Users] Compare to Skype
Eric ManxPower Wieling wrote: There are 2 issues here. 1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to transport audio for SIP (and other protocols). This means that ANY jitter on the SIP Phone - Asterisk link will cause audio problems. This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. The ideal scenario from a latency point of view is for the end points to handle jitter buffering. I use Polycom 500's with G711 over a path where jitter can be quite severe on occasion and they handle it very well. Although I have not tried them, one would expect Cisco's to work well also. Regards, Richard ___ --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
Re: [Asterisk-Users] Compare to Skype
This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. I agree with that. Asterisk should just forward any RTP immediately and let endpoints handle the jitter buffer - unless asterisk is the endpoint itself (e.g. with phones plugged in its fxs ports). ___ --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
Re: [Asterisk-Users] Compare to Skype
On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly: This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. I agree with that. Asterisk should just forward any RTP immediately and let endpoints handle the jitter buffer - unless asterisk is the endpoint itself (e.g. with phones plugged in its fxs ports). That makes sense if asterisk is just serving as a gateway, passing on audio to other machines, but if it's processing the audio on its own, that's not so good - it'd mess up recordings, for one. -- Jon-o Addleman - http://redowl.dyndns.org ___ --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
RE: [Asterisk-Users] Compare to Skype
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 [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Compare to Skype
[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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
RE: [Asterisk-Users] Compare to Skype
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 greatbut 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] Date: Sun, April 30, 2006 9:09 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 ___ --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
Re: [Asterisk-Users] Compare to Skype
[EMAIL PROTECTED] wrote: 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 greatbut 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. I am also a supporter of PSGW although on my AMD it never worked. Now it is getting obsolete at all, since I switch next week finally to a Linux desktop I never heard about Uplink, where is it, does it work? From the uplink web: System Requirements * Windows 98/2000/Me/XP/2003 sigh 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
Re: [Asterisk-Users] Compare to Skype
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? There are 2 issues here. 1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to transport audio for SIP (and other protocols). This means that ANY jitter on the SIP Phone - Asterisk link will cause audio problems. 2) Asterisk times it's outgoing audio based on the incoming audio. Therefore, if there is jitter on the SIP Phone - Asterisk link then Asterisk will replicate that jitter on the Asterisk - SIP Phone direction. REMEMBER, a jitter buffer only applies on INCOMING audio (from the standpoint of the device). These two issues are the main reason I have not deployed remote SIP phones for my clients. I believe that BOTH of these issues will be fixed in Asterisk 1.4.x, which should be released sometime this summer. -- Now accepting new clients in New Orleans, Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [Asterisk-Users] Compare to Skype
There are 2 issues here. 1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to transport audio for SIP (and other protocols). This means that ANY jitter on the SIP Phone - Asterisk link will cause audio problems. 2) Asterisk times it's outgoing audio based on the incoming audio. Therefore, if there is jitter on the SIP Phone - Asterisk link then Asterisk will replicate that jitter on the Asterisk - SIP Phone direction. REMEMBER, a jitter buffer only applies on INCOMING audio (from the standpoint of the device). These two issues are the main reason I have not deployed remote SIP phones for my clients. So, he should probably try an IAX softphone and see how that compare hth ___ --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
[Asterisk-Users] Compare to Skype
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 ___ --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
Re: [Asterisk-Users] Compare to Skype
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 ___ --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
Re: [Asterisk-Users] Compare to Skype
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 ___ --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
