Hi all,
Here is a scenario, i m working on, If anyone can comment on it.
Scenario: I have two SIP domains, abc.com and xvz.com sip.abc.com is running ser and has registered users with xxxx@abc.com similarly sip.xyz.com is serving users xxxx@xyz.com sip.abc.com and sip.xyz.com can be resolved through standard dns servers as SRV hosts for their respective domains. There is no explicitly defined routing procedure on ser.conf for other domain, in both SIP servers machines.
Is it possible for xxxx@abc.com to register with sip.abc.com and place a call to user xxxx@xyz.com. While trying, i m getting no packet on sip.abc.com not sip.xyz.com for the subject call.
Any comment?
JF
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Hello,
yes, it is possible. The default config file contains condition if (uri==myself) { do something };
t_relay();
What does that mean: If the Request-URI of a message contains a domain that belons to the server, the server does some processing like registration/unregistration etc..
If the Request-URI contains a domain that doesn't belong to the server then the server will simply forward it there, without any special processing.
So your setup should work providing that you have correct DNS records.
Jan.
On 04-10 00:19, John Foster wrote:
Hi all,
Here is a scenario, i m working on, If anyone can comment on it.
Scenario: I have two SIP domains, abc.com and xvz.com sip.abc.com is running ser and has registered users with xxxx@abc.com similarly sip.xyz.com is serving users xxxx@xyz.com sip.abc.com and sip.xyz.com can be resolved through standard dns servers as SRV hosts for their respective domains. There is no explicitly defined routing procedure on ser.conf for other domain, in both SIP servers machines.
Is it possible for xxxx@abc.com to register with sip.abc.com and place a call to user xxxx@xyz.com. While trying, i m getting no packet on sip.abc.com not sip.xyz.com for the subject call.
Any comment?
JF
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Hi,
But Jan, i m bit surprised that y SIP Phone directly contacts other domain, no packet comes on the sip.abc.com, but to make a call from xxxx@sip.abc.com to xxxx@sip.xyz.com SIP client(xxxx@sip.abc.com) directly contacts sip.xyz.com to get address of user xxxx@sip.xyz.com, is this due to protocol design? or any abnormality?
As the logic of if(uri==myself) {} else{} would match only when packets will come to ser. Isnt it?
JF
Jan Janak jan@iptel.org wrote: Hello,
yes, it is possible. The default config file contains condition if (uri==myself) { do something };
t_relay();
What does that mean: If the Request-URI of a message contains a domain that belons to the server, the server does some processing like registration/unregistration etc..
If the Request-URI contains a domain that doesn't belong to the server then the server will simply forward it there, without any special processing.
So your setup should work providing that you have correct DNS records.
Jan.
On 04-10 00:19, John Foster wrote:
Hi all,
Here is a scenario, i m working on, If anyone can comment on it.
Scenario: I have two SIP domains, abc.com and xvz.com sip.abc.com is running ser and has registered users with xxxx@abc.com similarly sip.xyz.com is serving users xxxx@xyz.com sip.abc.com and sip.xyz.com can be resolved through standard dns servers as SRV hosts for their respective domains. There is no explicitly defined routing procedure on ser.conf for other domain, in both SIP servers machines.
Is it possible for xxxx@abc.com to register with sip.abc.com and place a call to user xxxx@xyz.com. While trying, i m getting no packet on sip.abc.com not sip.xyz.com for the subject call.
Any comment?
JF
Do you Yahoo!? The New Yahoo! Shopping - with improved product search _______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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At 10:10 AM 10/6/2003, John Foster wrote:
Hi,
But Jan, i m bit surprised that y SIP Phone directly contacts other domain, no packet comes on the sip.abc.com, but to make a call from mailto:xxxx@sip.abc.comxxxx@sip.abc.com to mailto:xxxx@sip.xyz.comxxxx@sip.xyz.com SIP client(mailto:xxxx@sip.abc.comxxxx@sip.abc.com) directly contacts sip.xyz.com to get address of user mailto:xxxx@sip.xyz.comxxxx@sip.xyz.com, is this due to protocol design? or any abnormality?
There is no reason for a client to send an outoging request through its proxy, unless it is forced to do so by some kind of "outbound proxy" option. That's how SIP works.
As the logic of if(uri==myself) {} else{} would match only when packets will come to ser. Isnt it?
Indeed, uri==myself does not match packets which never hit SER.
-jiri
Hi Jiri,
Then what can be the way to get accounting of all calls at originating domain's ser? As if client removes outbound proxy from his IP Phone conf, and the carrier(ser operator) wants to account that user for those calls. (One way could be to force rtpproxy...right?) any other idea
JF
Jiri Kuthan jiri@iptel.org wrote: At 10:10 AM 10/6/2003, John Foster wrote:
Hi,
But Jan, i m bit surprised that y SIP Phone directly contacts other domain, no packet comes on the sip.abc.com, but to make a call from xxxx@sip.abc.com to xxxx@sip.xyz.com SIP client(xxxx@sip.abc.com) directly contacts sip.xyz.com to get address of user xxxx@sip.xyz.com, is this due to protocol design? or any abnormality?
There is no reason for a client to send an outoging request through its proxy, unless it is forced to do so by some kind of "outbound proxy" option. That's how SIP works.
As the logic of if(uri==myself) {} else{} would match only when packets will come to ser. Isnt it?
Indeed, uri==myself does not match packets which never hit SER.
-jiri
--------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search
John Foster writes:
Then what can be the way to get accounting of all calls at originating domain's ser? As if client removes outbound proxy from his IP Phone conf, and the carrier(ser operator) wants to account that user for those calls.
with sip and enum voip becomes a peer-to-peer internet service among many other such services. if a user subscribed to open internet service, there is no way anyone could account for the calls made by that user and it would be a big mistake to turn open internet into a new kind of closed phone network.
-- juha
:)account for, includes call logging, and analysis. Not commercial accounting or jailing a user
JF
Juha Heinanen jh@tutpro.com wrote: John Foster writes:
Then what can be the way to get accounting of all calls at originating domain's ser? As if client removes outbound proxy from his IP Phone conf, and the carrier(ser operator) wants to account that user for those calls.
with sip and enum voip becomes a peer-to-peer internet service among many other such services. if a user subscribed to open internet service, there is no way anyone could account for the calls made by that user and it would be a big mistake to turn open internet into a new kind of closed phone network.
-- juha
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John Foster writes:
:)account for, includes call logging, and analysis.
which you cannot do if the user susbcribes to internet service and want to use sip as a p2p application.
Not commercial accounting or jailing a user
i didn't say anything about these although call logging has in many cases been used for jailing.
what you can do is to account for calls that terminate at your own UAs, e.g. pstn gws.
-- juha
On 06-10 22:49, John Foster wrote:
Not commercial accounting or jailing a user
Yes, not at the beginning to attract new customers, but maybe later, who knows, the big brother is watching you :-)
Jan.
What would you like to account for if no service is provided in this case?
-jiri
At 06:53 AM 10/7/2003, John Foster wrote:
Hi Jiri,
Then what can be the way to get accounting of all calls at originating domain's ser? As if client removes outbound proxy from his IP Phone conf, and the carrier(ser operator) wants to account that user for those calls. (One way could be to force rtpproxy...right?) any other idea
JF
Jiri Kuthan jiri@iptel.org wrote: At 10:10 AM 10/6/2003, John Foster wrote:
Hi,
But Jan, i m bit surprised that y SIP Phone directly contacts other domain, no packet comes on the sip.abc.com, but to make a call from xxxx@sip.abc.com to xxxx@sip.xyz.com SIP client(xxxx@sip.abc.com) directly contacts sip.xyz.com to get address of user xxxx@sip.xyz.com, is this due to protocol design? or any abnormality?
There is no reason for a client to send an outoging request through its proxy, unless it is forced to do so by some kind of "outbound proxy" option. That's how SIP works.
As the logic of if(uri==myself) {} else{} would match only when packets will come to ser. Isnt it?
Indeed, uri==myself does not match packets which never hit SER.
-jiri
Do you Yahoo!? http://shopping.yahoo.com/?__yltc=s%3A150000443%2Cd%3A22708228%2Cslk%3Atext%2Csec%3AmailThe New Yahoo! Shopping - with improved product search
-- Jiri Kuthan http://iptel.org/~jiri/
Like if my network is being used for dialout only, bandwidth/cost analysis, future projections etc
Like if we see at VoIP sites they charge user on outgoings,by limiting them to their software they can force UAC to use proxy. A Call history can be maintained for VoIP to VoIP Calls..
I think its quite useful piece of information,...isnt it
JF
Jiri Kuthan jiri@iptel.org wrote: What would you like to account for if no service is provided in this case?
-jiri
At 06:53 AM 10/7/2003, John Foster wrote:
Hi Jiri,
Then what can be the way to get accounting of all calls at originating domain's ser? As if client removes outbound proxy from his IP Phone conf, and the carrier(ser operator) wants to account that user for those calls. (One way could be to force rtpproxy...right?) any other idea
JF
Jiri Kuthan wrote: At 10:10 AM 10/6/2003, John Foster wrote:
Hi,
But Jan, i m bit surprised that y SIP Phone directly contacts other domain, no packet comes on the sip.abc.com, but to make a call from xxxx@sip.abc.com to xxxx@sip.xyz.com SIP client(xxxx@sip.abc.com) directly contacts sip.xyz.com to get address of user xxxx@sip.xyz.com, is this due to protocol design? or any abnormality?
There is no reason for a client to send an outoging request through its proxy, unless it is forced to do so by some kind of "outbound proxy" option. That's how SIP works.
As the logic of if(uri==myself) {} else{} would match only when packets will come to ser. Isnt it?
Indeed, uri==myself does not match packets which never hit SER.
-jiri
Do you Yahoo!? The New Yahoo! Shopping - with improved product search
-- Jiri Kuthan http://iptel.org/~jiri/
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At 10:45 AM 10/7/2003, John Foster wrote:
Like if my network is being used for dialout only, bandwidth/cost analysis, future projections etc
Like if we see at VoIP sites they charge user on outgoings,by limiting them to their software they can force UAC to use proxy. A Call history can be maintained for VoIP to VoIP Calls..
I think its quite useful piece of information,...isnt it
I'm not sure -- if you are interested in bandwith analysis, analyse bandwith.
If subsribers see a value in using an outbound proxy (say for web-maintained list of outbound calls), they can turn on outbound-proxy option in their UAC.
-jiri