Re: [Asterisk-Users] g729 and latency measures
On Mon, 2006-03-20 at 11:38 +0530, ram wrote: Hi what is mtr ? where can i find that http://www.google.com/linux?hl=enlr=q=mtrbtnG=Search Pete ___ --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] g729 and latency measures
Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --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] g729 and latency measures
Thanks Rich, but i'm only allowed to use g729. you said that some folks run high latency connections, but is 300ms high in my setup? On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] g729 and latency measures
Yes, 300ms seems very high if there is no satellite link involved. g729 should be just fine if that's what you're stuck with. Erick Perez wrote: Thanks Rich, but i'm only allowed to use g729. you said that some folks run high latency connections, but is 300ms high in my setup? On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? The objective should be to keep latency as low as possible, however some folks do run asterisk via satellite which as a very lengthy latency. How can I measure this latency all the way to the asterisk? Several ways depending on how accurate a measurement you want. A simple ping would give a starting point. A much more expensive way is to use VoIP analysis software to measure it, but be prepared to spend at least $1,500 (US) to do that. Should I ping from SITE_A to the asterisk box with 8k packets? If you want to emulate a sip/iax packet, use a packet size of about 200 bytes. If I can't install ADSL for the moment, will the above setup work? Probably a bigger issue to address relates to what other traffic might be passing across the dsl and/or wireless channel that might be consuming bandwidth and impacting the rtp packets. Broadcasts originating from devices outside your control (other isp users), hackers attempting to access your ip addresses (at both ends), data traffic between your two endpoints, etc, are just some thoughts of items using a portion of the bandwidth available. Might also think about jitter (eg, variations in latency) and what that might do to your end to end communications. There are other low bandwidth codecs available that could be used instead of g729. Some include ilbc, g726, gsm, etc. Each consumes different bandwidths, and each provide a slightly different quality of audio. See the wiki for more detail on what each consumes for bandwidth on the wire. ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] g729 and latency measures
Erick Perez wrote: How can I measure this latency all the way to the asterisk? I have found two good ways to monitor routes for VOIP. Install mtr and run mtr your.voipserver to find where you are seeing the latency, and then install smokeping (not so easy to install) and you will be able to monitor the latency over time. I find smokeping the most reliable way to visually gauge route quality. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] g729 and latency measures
Hi what is mtr ? where can i find that ram On 3/20/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Erick Perez wrote: How can I measure this latency all the way to the asterisk? I have found two good ways to monitor routes for VOIP. Install mtr andrun mtr your.voipserver to find where you are seeing the latency, andthen install smokeping (not so easy to install) and you will be able to monitor the latency over time. I find smokeping the most reliable way tovisually gauge route quality.--Chris MasonNetConcepts(264) 497-5670 Fax: (264) 497-8463Int:(305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This message has been scanned for viruses anddangerous content by MailScanner, and is believed to be clean.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
