Hello,
I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients.
How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying?
Look forward to your answers...
JB
Hi Jerlique!
It all depends on requirements: - how many clients - do you want to operate your own PSTN gateway - are the clients behind NAT - ...
For example, if this is for a small office (20 clients sitting in an office, existing ISDN lines ...), then Asterisk alone will handle this - no need for ser.
If you are thinking of an ITSP setup with >= 1000 clients, then you will need a SIP proxy (e.g. ser) for user registration, call routing and NAT traversal. For PSTN connectivity you will have several PSTN Gateways (Asterisk or commercial ones) or you buy PSTN connectivity from a terminatione provider (nufone, level3, globalcrossing, ...).
regards, klaus
Jerlique Ban wrote:
Hello,
I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients.
How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying?
Look forward to your answers...
JB
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi
Depends on what you really want to do, and how you wish to scale, several setups which I have been invloved with use a combo of SER and asterisk, asterisk is great for alot of pbx functionality, but to be honest SER and asterisk can be awkward to get right all the time, passing mof message back and forth etc etc.
SER is a great proxy, and can handle all the registrations, althought it would be nice if there were a nice clustering solution for it, which worked with NAT, but i am sure that will come along soon enough, even with 3rd party apps like LVS. But even as a stand alone it can handle alot of queries, but as with nething fine tuning is time-based process and you cannot really build it to perfection on day one.
Asterisk can also handle gateway functionaility, and its a personal choice if you opt for oneof these or cisco box or something. carrier grade is not a easy option, and cannot be built overnight, but thats the great thing "build it and they will come" :-).
Iqbal
Jerlique Ban wrote:
Hello,
I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients.
How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying?
Look forward to your answers...
JB
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
Vonage is offering an unlimited USA & Canada, how are they able to do this?
On 6/22/05, Iqbal iqbal@gigo.co.uk wrote:
Hi
Depends on what you really want to do, and how you wish to scale, several setups which I have been invloved with use a combo of SER and asterisk, asterisk is great for alot of pbx functionality, but to be honest SER and asterisk can be awkward to get right all the time, passing mof message back and forth etc etc.
SER is a great proxy, and can handle all the registrations, althought it would be nice if there were a nice clustering solution for it, which worked with NAT, but i am sure that will come along soon enough, even with 3rd party apps like LVS. But even as a stand alone it can handle alot of queries, but as with nething fine tuning is time-based process and you cannot really build it to perfection on day one.
Asterisk can also handle gateway functionaility, and its a personal choice if you opt for oneof these or cisco box or something. carrier grade is not a easy option, and cannot be built overnight, but thats the great thing "build it and they will come" :-).
Iqbal
Jerlique Ban wrote:
Hello,
I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients.
How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying?
Look forward to your answers...
JB
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
how unlimited is unlimited
Iqbal
On 7/7/2005, "Budi Gautama" budi.gautama@gmail.com wrote:
Vonage is offering an unlimited USA & Canada, how are they able to do this?
On 6/22/05, Iqbal iqbal@gigo.co.uk wrote:
Hi
Depends on what you really want to do, and how you wish to scale, several setups which I have been invloved with use a combo of SER and asterisk, asterisk is great for alot of pbx functionality, but to be honest SER and asterisk can be awkward to get right all the time, passing mof message back and forth etc etc.
SER is a great proxy, and can handle all the registrations, althought it would be nice if there were a nice clustering solution for it, which worked with NAT, but i am sure that will come along soon enough, even with 3rd party apps like LVS. But even as a stand alone it can handle alot of queries, but as with nething fine tuning is time-based process and you cannot really build it to perfection on day one.
Asterisk can also handle gateway functionaility, and its a personal choice if you opt for oneof these or cisco box or something. carrier grade is not a easy option, and cannot be built overnight, but thats the great thing "build it and they will come" :-).
Iqbal
Jerlique Ban wrote:
Hello,
I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients.
How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying?
Look forward to your answers...
JB
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
It must be sutid, but in this Invite, where is the Request-URI, is it the on in the Invite itself, of another header?
INVITE sip:5263983227321073@212.23.62.147:5060 SIP/2.0..Record-Route: sip:82.146.123.252:5070;ftag=9073921E58511927264;lr=on..Via: SIP/2.0/UDP 82.146.123.252:5070;branch= z9hG4bK762f.696c43e1.0..Via: SIP/2.0/UDP 82.146.123.252:5065..To: sip:3227321073@82.146.123.252;user=phone..From: Oliviersip:3237470305@82.146.123.252;user=phone;tag=90 73921E58511927264..Call-ID: 9ccddde28dd1cc47cafcf074e26b4e74@82.146.123.252..CSeq: 2 INVITE..Proxy-Authorization: DIGEST algorithm=MD5,nonce="42cd00e55eed3e7b6fabd16393bbd0
4ca78816c1",realm="finalcut.be",response="8839f490e49df69ba0047cb246c2347e", uri="sip:027321073@82.146.123.252",username="3237470305"..Max-Forwards: 15..Contact: <sip:323747 0305@82.146.123.252:5065;user=phone>..Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER..Supported: replaces..User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone..Cont ent-Type: application/sdp..Content-Length: 299....v=0..o=TelogyUnknown0000 1928256 1928256 IN IP4 192.168.2.107..s=RTP Audio..c=IN IP4 82.146.123.252..t=0 0..m=audio 36246 RTP/AVP 18 0 8 101..a=rtpmap:18 G729/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=direction:active..a=nortpproxy: yes..
Thanks,
Olivier
The INVITE you posted has a R-URI of
sip:5263983227321073@212.23.62.147:5060
which is on the [very] first line of the message. In other words, this is the request URI
INVITE sip:5263983227321073@212.23.62.147:5060 SIP/2.0
Regards, Paul
On 7/7/05, Olivier Taylor olivier.taylor@gmail.com wrote:
It must be sutid, but in this Invite, where is the Request-URI, is it the on in the Invite itself, of another header?
INVITE sip:5263983227321073@212.23.62.147:5060 SIP/2.0..Record-Route: sip:82.146.123.252:5070;ftag=9073921E58511927264;lr=on..Via: SIP/2.0/UDP 82.146.123.252:5070;branch= z9hG4bK762f.696c43e1.0..Via: SIP/2.0/UDP 82.146.123.252:5065..To: sip:3227321073@82.146.123.252;user=phone..From: Oliviersip:3237470305@82.146.123.252;user=phone;tag=90 73921E58511927264..Call-ID: 9ccddde28dd1cc47cafcf074e26b4e74@82.146.123.252..CSeq: 2 INVITE..Proxy-Authorization: DIGEST algorithm=MD5,nonce="42cd00e55eed3e7b6fabd16393bbd0
4ca78816c1",realm="finalcut.be",response="8839f490e49df69ba0047cb246c2347e", uri="sip:027321073@82.146.123.252",username="3237470305"..Max-Forwards: 15..Contact: <sip:323747 0305@82.146.123.252:5065;user=phone>..Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER..Supported: replaces..User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone..Cont ent-Type: application/sdp..Content-Length: 299....v=0..o=TelogyUnknown0000 1928256 1928256 IN IP4 192.168.2.107..s=RTP Audio..c=IN IP4 82.146.123.252..t=0 0..m=audio 36246 RTP/AVP 18 0 8 101..a=rtpmap:18 G729/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=direction:active..a=nortpproxy: yes..
Thanks,
Olivier
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hello
if you are a telco and/or have agreements with other telcos, you can do this. i.e. here in Brazil 2 phone companies don't pay a penny to each other to terminate calls on the other's networks (Vésper and Telefônica). in US it is far more common
Cheers !3runo
Iqbal wrote:
how unlimited is unlimited
Iqbal
On 7/7/2005, "Budi Gautama" budi.gautama@gmail.com wrote:
Vonage is offering an unlimited USA & Canada, how are they able to do this?
Is there any limit in the no of minutes of Vonage/BroadVoice UNLIMITED serives ?
On 7/7/05, Bruno Lopes F. Cabral bruno@openline.com.br wrote:
Hello
if you are a telco and/or have agreements with other telcos, you can do this. i.e. here in Brazil 2 phone companies don't pay a penny to each other to terminate calls on the other's networks (Vésper and Telefônica). in US it is far more common
Cheers !3runo
Iqbal wrote:
how unlimited is unlimited
Iqbal
On 7/7/2005, "Budi Gautama" budi.gautama@gmail.com wrote:
Vonage is offering an unlimited USA & Canada, how are they able to do this?
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I've heard reports that voicepulse cuts you off and never answer your complains/don't let you sign up again if you pass 2k minutes/month. I never heard anything about vonage following this same policy, though, so YMMV.
Budi Gautama wrote:
Is there any limit in the no of minutes of Vonage/BroadVoice UNLIMITED serives ?
if you are a telco and/or have agreements with other telcos, you can do this.
-
Bruno Lopes F. Cabral -
Budi Gautama -
Iqbal -
Jerlique Ban -
Klaus Darilion -
Olivier Taylor -
Paul Hazlett