Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers. It depends on how you did your tests.. If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic.. Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading.. Later.. Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip bandwidth usage
This is exactly what I did. I used Xten's GSM driver to call a Zap extension. Readings where 100 Kbits/s. Using uLAW returned 80 Kbits/s !!! I also downloaded Xten pro to test their g729 codec, readings were even worse. That's why I'm so intrigued. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: quinta-feira, 30 de outubro de 2003 10:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip bandwidth usage Paulo Mannheimer wrote: That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers. It depends on how you did your tests.. If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic.. Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading.. Later.. Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: This is exactly what I did. I used Xten's GSM driver to call a Zap extension. Readings where 100 Kbits/s. Using uLAW returned 80 Kbits/s !!! I also downloaded Xten pro to test their g729 codec, readings were even worse. That's why I'm so intrigued. That is odd.. Especially since you got higher bandwidth usage with GSM than you did with G.711.. This is a good site to give you an indication of the bandwidth requirements for various codecs under various conditions.. http://www.packetizer.com/iptel/bandcalc.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip bandwidth usage
Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
