1. We would like to transcode from g711mu internally to g729 for PSTN calls. Is this possible? 2. We are using sipX as an open source PBX. We'd like to use FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom phones use SIP REFER messages to do call transfers and our ITSPs can't handle those. So, we've tried the "under the hood" FreeSwitch within sipX. It works for many applications but we need to scale a unified system for larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don't plan to point any internal phones to the FreeSwitch box (we're using sipX as an internal PBX). 3. If we use multiple ITSPs, can we assign a priority to each one, so one will be chosen first? 4. I understand no performance metrics are posted. Can anyone share their experiences with passing RTP through the server? Does RTP affect performance in a big way (very open ended question, I know). We are looking for ball park answers so we'll have some idea of scalability for our designs. Thanks, Ken Fulmer
If you plan to transcode g729 to g711 it will impact performance dramatically but you should still get a few hundered channels up on a nice server box. You could contact consulting@freeswitch.org about g729 license options. With no g729 you could scale a nice 8 core box into a few thousand channels pushing rtp. Do you have a requirement to transcode g729? you could just use g711 right from the phone couldn't you? On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer < kenfulmer@icstechnologysolutions.com> wrote:
1. We would like to transcode from g711mu internally to g729 for PSTN calls. Is this possible?
2. We are using sipX as an open source PBX. We’d like to use FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom phones use SIP REFER messages to do call transfers and our ITSPs can’t handle those. So, we’ve tried the “under the hood” FreeSwitch within sipX. It works for many applications but we need to scale a unified system for larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don’t plan to point any internal phones to the FreeSwitch box (we’re using sipX as an internal PBX).
3. If we use multiple ITSPs, can we assign a priority to each one, so one will be chosen first?
4. I understand no performance metrics are posted. Can anyone share their experiences with passing RTP through the server? Does RTP affect performance in a big way (very open ended question, I know). We are looking for ball park answers so we’ll have some idea of scalability for our designs.
Thanks,
Ken Fulmer
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Thanks for your reply. If we only want to route calls from one provider to another or from the PSTN to an internal PBX (and vice versa), would we use the external SIP profiles for each "trunk"? Thanks, Ken Fulmer From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, March 19, 2010 12:22 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie Questions If you plan to transcode g729 to g711 it will impact performance dramatically but you should still get a few hundered channels up on a nice server box. You could contact consulting@freeswitch.org about g729 license options. With no g729 you could scale a nice 8 core box into a few thousand channels pushing rtp. Do you have a requirement to transcode g729? you could just use g711 right from the phone couldn't you? On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer <kenfulmer@icstechnologysolutions.com> wrote: 1. We would like to transcode from g711mu internally to g729 for PSTN calls. Is this possible? 2. We are using sipX as an open source PBX. We'd like to use FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom phones use SIP REFER messages to do call transfers and our ITSPs can't handle those. So, we've tried the "under the hood" FreeSwitch within sipX. It works for many applications but we need to scale a unified system for larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don't plan to point any internal phones to the FreeSwitch box (we're using sipX as an internal PBX). 3. If we use multiple ITSPs, can we assign a priority to each one, so one will be chosen first? 4. I understand no performance metrics are posted. Can anyone share their experiences with passing RTP through the server? Does RTP affect performance in a big way (very open ended question, I know). We are looking for ball park answers so we'll have some idea of scalability for our designs. Thanks, Ken Fulmer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don’t plan to point any internal phones to the FreeSwitch box (we’re using sipX as an internal PBX).
The "internal" and "external" monikers are arbitrary names given to two SIP profiles. It's completely possible to run a FreeSWITCH box with a single profile, named internal, external, or general, or whatever. The samples are provided and so-labelled because one of the biggest problems new users have with PBXs is distinguishing inbound and outbound calls. If an inbound call somehow gets into an outbound dialplan, you can end up with a situation where someone starts using your PBX to make calls, without your consent. FreeSWITCH, by permitting association of ports and/or IP addresses with specific SIP profiles makes it trivial to keep these calls completely segregated, making it a little harder to do "stupid" things. -jbn
I'm not sure I understand. Aren't the sip profiles where I define the ip address of my internal PBX and / or an upstream carrier? If so, how can I only use one? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Justin B Newman Sent: Friday, March 19, 2010 2:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie Questions
larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don't plan to point any internal phones to the FreeSwitch box (we're using sipX as an internal PBX).
The "internal" and "external" monikers are arbitrary names given to two SIP profiles. It's completely possible to run a FreeSWITCH box with a single profile, named internal, external, or general, or whatever. The samples are provided and so-labelled because one of the biggest problems new users have with PBXs is distinguishing inbound and outbound calls. If an inbound call somehow gets into an outbound dialplan, you can end up with a situation where someone starts using your PBX to make calls, without your consent. FreeSWITCH, by permitting association of ports and/or IP addresses with specific SIP profiles makes it trivial to keep these calls completely segregated, making it a little harder to do "stupid" things. -jbn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer <kenfulmer@icstechnologysolutions.com> wrote:
I'm not sure I understand. Aren't the sip profiles where I define the ip address of my internal PBX and / or an upstream carrier? If so, how can I only use one?
A SIP profile associates an IP address and port with a series of settings and with a series of gateways. In so much as the gateways are defined properly, I'm not aware of an inherent need to segregate them into two SIP profiles. That said, having incoming and outgoing calls going through the same SIP profile significantly increases the opportunity for error, which could result in security vulnerabilities. -jbn
Ok, thanks. I think I'm beginning to understand a little more clearly. Ken -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Justin B Newman Sent: Monday, March 22, 2010 9:29 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie Questions On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer <kenfulmer@icstechnologysolutions.com> wrote:
I'm not sure I understand. Aren't the sip profiles where I define the ip address of my internal PBX and / or an upstream carrier? If so, how can I only use one?
A SIP profile associates an IP address and port with a series of settings and with a series of gateways. In so much as the gateways are defined properly, I'm not aware of an inherent need to segregate them into two SIP profiles. That said, having incoming and outgoing calls going through the same SIP profile significantly increases the opportunity for error, which could result in security vulnerabilities. -jbn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sorry for the repeat message. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Justin B Newman Sent: Monday, March 22, 2010 9:29 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie Questions On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer <kenfulmer@icstechnologysolutions.com> wrote:
I'm not sure I understand. Aren't the sip profiles where I define the ip address of my internal PBX and / or an upstream carrier? If so, how can I only use one?
A SIP profile associates an IP address and port with a series of settings and with a series of gateways. In so much as the gateways are defined properly, I'm not aware of an inherent need to segregate them into two SIP profiles. That said, having incoming and outgoing calls going through the same SIP profile significantly increases the opportunity for error, which could result in security vulnerabilities. -jbn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Anthony Minessale
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Justin B Newman
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Ken Fulmer