Hi,
I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VMs (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish.
Any suggestions or guides would be immensely appreciated.
Thank you all for your time.
Kevin Farrell Pelletier - Technicien informatique
TI // Réseautique // Téléphonie IP // Programmation IT // Networking // VoIP // Application development
9060, boul Parkway, Anjou, Québec, Canada H1J1N5 Téléphone / Phone : 514.907.2000 Ext.203 F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : http://www.connextek.ca/ www.connextek.ca Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire Consider the ENVIRONMENT before printing
Hello Kevin,If I understood properly you want to build a system which authenticates users and routes the Asterisk servers for communication. First, Kamailio supports the routing, balancing and authentication. For example we use Kamailio and Freeswitch. Here the how its work:We have 1 Kamailio server that makes routes and balancing issues.First client goes to our Kamailio servers: Client -> Internet -> Kamailio (authentication) (address, asked for communication) After that, Kamailio looks the Freeswitch servers, which is free for routing. (sending)Kamailio -----------------> Freeswitch Server (user req) After routing proccess, Kamailio fade from the scene and clients start communicate with themselves via Freeswitch servers. BTW, our Freeswitch servers and Kamailio servers stay on different servers. Of course you can serve on same server too.If I understood properly, you can do it like this. If I did not, you can give more details for understanding :) Regards. Barış.From: kfpelletier@connextek.ca To: sr-users@lists.sip-router.org; sr-dev@lists.sip-router.org; buisness@lists.kamailio.org Date: Fri, 26 Feb 2016 15:35:50 -0500 Subject: [SR-Users] Help
Hi,
I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish.
Any suggestions or guides would be immensely appreciated.
Thank you all for your time.
Kevin Farrell Pelletier - Technicien informatique
TI // Réseautique // Téléphonie IP // Programmation
IT // Networking // VoIP // Application development
9060, boul Parkway, Anjou, Québec, Canada H1J1N5
Téléphone / Phone : 514.907.2000 Ext.203
F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : www.connextek.ca
Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire
Consider the ENVIRONMENT before printing
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I think the best guide closest to your description is here : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Here is what you need to do. (*Besides mentioning what you tried and what problems were faced*).
1 - Configure kamailio to use the DB schema where your users are stored with their password and PBX to use info. 2 - Point Phones to REGISTER to your kamailio. 3 - When a User makes a call execute query in Kamailio to find what client this user belongs to and what Asterisk it should be routed to. 4 - Send Calls to the selected Asterisk.
You can further add Memcache to save your DB query in step-3.
Good Part: 1 - Kamailio Authenticates all users and calls. 2 - One Public IP to point all domains/PBX tenants to. 3 - Use RTPproxy to bridge media to Asterisks and you can shift your Asterisks on Private Subnet too.(depends on your design) 4 - Sending a hand crafted REGISTER to Asterisk makes asterisk aware of the device state and hence BLF/MWI are handled by Asterisk.
Less Good Part: (As I see it)
Kamailio sends REGISTER packet to just one Asterisk ! thereby only one server out of pool is aware of the device states. It can be resolved by extra effort required as following:
a) Yes we can use Dispatcher and send to failover/loadbalanced asterisks in the pool b) A script of some sorts can be written and started in asterisk servers to share device states/hints and \ hence all asterisk servers in pool know whats going on. (I haven't tried it myself) c) REGFWD route can be blocked and BLF, MWI are handled solely by Kamailio. (* I personally had rough time with this mostly due to different standards *from IP Phones)
I'd love to hear other valuable suggestions and experiences.
Regards, Sammy Hello Kevin, If I understood properly you want to build a system which authenticates users and routes the Asterisk servers for communication.
First, Kamailio supports the routing, balancing and authentication. For example we use Kamailio and Freeswitch. Here the how its work: We have 1 Kamailio server that makes routes and balancing issues. First client goes to our Kamailio servers:
*Client -> Internet -> Kamailio (authentication) (address, asked for communication)*
After that, Kamailio looks the Freeswitch servers, which is free for routing.
* (sending)* *Kamailio -----------------> Freeswitch Server* * (user req)*
After routing proccess, Kamailio fade from the scene and clients start communicate with themselves via Freeswitch servers.
BTW, our Freeswitch servers and Kamailio servers stay on different servers. Of course you can serve on same server too. If I understood properly, you can do it like this. If I did not, you can give more details for understanding :)
Regards.
Barış. ------------------------------ From: kfpelletier@connextek.ca To: sr-users@lists.sip-router.org; sr-dev@lists.sip-router.org; buisness@lists.kamailio.org Date: Fri, 26 Feb 2016 15:35:50 -0500 Subject: [SR-Users] Help
Hi,
I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish.
Any suggestions or guides would be immensely appreciated.
Thank you all for your time.
*Kevin Farrell Pelletier - Technicien informatique*
TI* //* Réseautique // Téléphonie IP *//* Programmation IT // Networking // VoIP // Application development
9060, boul Parkway, Anjou, Québec, Canada H1J1N5 * Téléphone / Phone :* 514.907.2000 Ext.203 * F :* 1.888.582.4001 - *SF/TF :* 1.855.907.2001* Web : *www.connextek.ca
* Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire * * Consider the ENVIRONMENT before printing*
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi
I found this info very helpful when setting up a scenario similar to your required setup.
https://txlab.wordpress.com/2012/10/21/kamailio-as-a-pass-through-proxy/
Gerry Kernan
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan@infinityit.ie
Managed IT Services Infinity IT - www.infinityit.ie IP Telephony Asterisk Consulting – www.asteriskconsulting.com Contact Centre Total Interact – www.totalinteract.com
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of SamyGo Sent: Sunday 28 February 2016 15:51 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List sr-users@lists.sip-router.org Subject: Re: [SR-Users] Help
Hi, I think the best guide closest to your description is here : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Here is what you need to do. (Besides mentioning what you tried and what problems were faced). 1 - Configure kamailio to use the DB schema where your users are stored with their password and PBX to use info. 2 - Point Phones to REGISTER to your kamailio. 3 - When a User makes a call execute query in Kamailio to find what client this user belongs to and what Asterisk it should be routed to. 4 - Send Calls to the selected Asterisk. You can further add Memcache to save your DB query in step-3. Good Part: 1 - Kamailio Authenticates all users and calls. 2 - One Public IP to point all domains/PBX tenants to. 3 - Use RTPproxy to bridge media to Asterisks and you can shift your Asterisks on Private Subnet too.(depends on your design) 4 - Sending a hand crafted REGISTER to Asterisk makes asterisk aware of the device state and hence BLF/MWI are handled by Asterisk. Less Good Part: (As I see it) Kamailio sends REGISTER packet to just one Asterisk ! thereby only one server out of pool is aware of the device states. It can be resolved by extra effort required as following: a) Yes we can use Dispatcher and send to failover/loadbalanced asterisks in the pool b) A script of some sorts can be written and started in asterisk servers to share device states/hints and \ hence all asterisk servers in pool know whats going on. (I haven't tried it myself) c) REGFWD route can be blocked and BLF, MWI are handled solely by Kamailio. ( I personally had rough time with this mostly due to different standards from IP Phones) I'd love to hear other valuable suggestions and experiences. Regards, Sammy Hello Kevin, If I understood properly you want to build a system which authenticates users and routes the Asterisk servers for communication.
First, Kamailio supports the routing, balancing and authentication. For example we use Kamailio and Freeswitch. Here the how its work: We have 1 Kamailio server that makes routes and balancing issues. First client goes to our Kamailio servers:
Client -> Internet -> Kamailio (authentication) (address, asked for communication)
After that, Kamailio looks the Freeswitch servers, which is free for routing.
(sending) Kamailio -----------------> Freeswitch Server (user req)
After routing proccess, Kamailio fade from the scene and clients start communicate with themselves via Freeswitch servers.
BTW, our Freeswitch servers and Kamailio servers stay on different servers. Of course you can serve on same server too. If I understood properly, you can do it like this. If I did not, you can give more details for understanding :)
Regards.
Barış.
From: kfpelletier@connextek.ca To: sr-users@lists.sip-router.org; sr-dev@lists.sip-router.org; buisness@lists.kamailio.org Date: Fri, 26 Feb 2016 15:35:50 -0500 Subject: [SR-Users] Help Hi,
I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish.
Any suggestions or guides would be immensely appreciated.
Thank you all for your time.
Kevin Farrell Pelletier - Technicien informatique
TI // Réseautique // Téléphonie IP // Programmation IT // Networking // VoIP // Application development
9060, boul Parkway, Anjou, Québec, Canada H1J1N5 Téléphone / Phone : 514.907.2000 Ext.203 F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : www.connextek.ca Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire Consider the ENVIRONMENT before printing
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Barış Şekerciler -
gerry kernan -
Kevin Farrell-Pelletier -
SamyGo