22 Jun
2005
22 Jun
'05
8:49 a.m.
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
22 Jun
22 Jun
10:20 a.m.
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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:11 p.m.
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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
.
7 Jul
7 Jul
4:10 a.m.
Vonage is offering an unlimited USA & Canada, how are they able to do this?
On 6/22/05, Iqbal <iqbal(a)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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
1:17 p.m.
how unlimited is unlimited
Iqbal
On 7/7/2005, "Budi Gautama" <budi.gautama(a)gmail.com> wrote:
Vonage is offering an unlimited USA & Canada, how
are they able to do this?
On 6/22/05, Iqbal <iqbal(a)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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
1:32 p.m.
New subject: [Serusers] Difference between URI and RURI in an Invite?
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:
Olivier<sip: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
7 p.m.
New subject: [Serusers] Difference between URI and RURI in an Invite?
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(a)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:
Olivier<sip: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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:05 p.m.
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(a)gmail.com> wrote:
>Vonage is offering an unlimited USA & Canada, how are they able to do this?
11 Jul
11 Jul
7:48 a.m.
Is there any limit in the no of minutes of Vonage/BroadVoice UNLIMITED serives ?
On 7/7/05, Bruno Lopes F. Cabral <bruno(a)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(a)gmail.com> wrote:
>Vonage is offering an unlimited USA & Canada, how are they able to do this?
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:30 p.m.
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.
7077
days inactive
7096
days old
9 comments
7 participants
participants (7)
-
Bruno Lopes F. Cabral -
Budi Gautama -
Iqbal -
Jerlique Ban -
Klaus Darilion -
Olivier Taylor -
Paul Hazlett