[Asterisk-Users] Asterisk without RTP streams vs. SER in statefull mode
I am am starting up a residential VOIP company and will be using only SIP both with my customers and my termination provider. I am also doing peering with other SIP networks. My goal is to be able to avoid handling the RTP streams as much as possible as to save bandwidth. My original Idea was to use SER to handle the Call routing and only use Asterisk for voicemail, forwarding, etc. However, when talking to a termination provider, he suggested that I could get the same results using asterisk by itself, by setting reinvite=yes. I was wondering what the comparative performance would be. Most of the comparisons I've seen have been comparing Asterisk in B2BUA mode ( handling RTP streams) to SER in stateless proxy mode (just relaying SIP transactions). So they have been comparing apples to oranges (Asterisk in it's slowest mode, with SER in its fastest). To get a true apples to apples comparison, SER would need to use functions such as t_relay() and loose_route() and probably a few more, which impede optimal performance; Asterisk would be configured with reinvite=yes (or is it canreinvite=yes?) so that it doesn't have the overhead of handling RTP and thus would have a higher performance. With all that said, what would the performance difference be? How many registrations/calls can I handle per machine (3.8 GHz P4 1GB RAM)? What are the advantages and disadvantages of each approach? ___ --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] Asterisk and RTP streams (just bumping)
Bumping, just in case it got lost in the shuffle today... I think this is an important thing to be able to do. Subject: [Asterisk-Users] Asterisk and RTP streams Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and RTP streams (just bumping)
Just as a FYI, the rtp port that is used for any device (asterisk or a sip adapter) is choosen by the device. If a sip adapter as an example attempts to initiate an rtp session, changing asterisk won't have any impact. Or, saying this a little different, asterisk can initiate an rtp session using an rtp/udp port from within the range specified in rtp.conf, but the other end of that rtp session (sip adapter) gets to pick its own udp port for the return data. The udp port selected is often times picked from a range that you have control over, but in all cases that I can think of, you can't tell the device to always pick a single predetermined udp port. So, unless I've really missed the point in the discussion below, I don't believe the proposed port selection changes can be accomplished. Bumping, just in case it got lost in the shuffle today... I think this is an important thing to be able to do. Subject: [Asterisk-Users] Asterisk and RTP streams Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and RTP streams (just bumping)
Thanks for the heads up. I guess BOTH devices then have to be configured to use the same port for outgoing as the incoming came in on. I know some UA's have this option, I'll just have to figure it out. Cheers, Sherwood --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Rich Adamson -Sent: Saturday, October 01, 2005 9:21 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: RE: [Asterisk-Users] Asterisk and RTP streams (just bumping) - -Just as a FYI, the rtp port that is used for any device -(asterisk or a sip adapter) is choosen by the device. If a -sip adapter as an example attempts to initiate an rtp -session, changing asterisk won't have any impact. - -Or, saying this a little different, asterisk can initiate an -rtp session using an rtp/udp port from within the range -specified in rtp.conf, but the other end of that rtp session -(sip adapter) gets to pick its own udp port for the return -data. The udp port selected is often times picked from a -range that you have control over, but in all cases that I can -think of, you can't tell the device to always pick a single -predetermined udp port. - -So, unless I've really missed the point in the discussion -below, I don't believe the proposed port selection changes -can be accomplished. - - - - Bumping, just in case it got lost in the shuffle today... I -think this - is an important thing to be able to do. - - Subject: [Asterisk-Users] Asterisk and RTP streams - - Guys, I've been poking around trying to find a good answer for this - via voip-info, google, etc... Haven't found anything that helps, so - maybe you mates could. - - A lot of my customers are using Linksys UAs (router/ATA -PAP2) and some - using Sipura SPA-2002s. Every once in a while, the customer -will get - one-way audio. I've read that this is commonly caused by -the outgoing - RTP port not being the same as the incoming RTP port. A lot -of other - devices (I found info on forcing Xten to do it) can be -forced to use - the same port for both, but these devices don't have an -option (that - I've been able to find, even in the provisioning configs) -to do this. So, my question is two-fold: - - 1. Can Asterisk be told to send the RTP stream for incoming and - outgoing always on the same set of ports? - 2. Does anyone know something that I'm missing for the -above mentioned - devices? They're all the 2 line version of the ATA and/or router - configs (wireless and wired) - - Thank you all in advance for your thoughts and comments. I -apologize - in advance if I missed something that was publicly available. - - Sherwood McGowan - - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - End of Original Message- - - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and RTP streams (just bumping)
Sherwood McGowan wrote: Bumping, just in case it got lost in the shuffle today... I think this is an important thing to be able to do. Subject: [Asterisk-Users] Asterisk and RTP streams Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) If you turn on nat=yes this will affect both SIP and RTP. Asterisk will then send to the same address as we receive RTP from, this is called symmetric RTP. THere's no way we can affect the address port range that the device tell us to send to, but we can ignore that in the case there is a NAT in between and send to whatever address the device sends audio from. The RTP port address we receive RTP on *from* the device is settable in rtp.conf. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and RTP streams
Sherwood, I have never known the RTP audio to be on only one port in sip. I believe it's always on 2. The one way audio is always a nat/firewall problem in sip. Sherwood McGowan wrote: Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and RTP streams
Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and rtp:// streams
I have Obsequium running and have developed a way to parse the .PLS files that it returns. Is there a way in Asterisk to play rtp:// streams as MOH? Thanks. -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
