I sent this earlier and got no responses. Perhaps this is not the right forum to ask this question. Can any one suggest a better place to go for this information?
Thanks, Bob
-----Original Message----- From: Bob Carlson Sent: Wednesday, November 03, 2004 3:22 PM To: 'SerUsers' Subject: Transfer and Conferencing
Let me apologize in advance for my question, which is a little rudimentary. We are just starting a project that will use SER and I am being forced to document right now how transfer and conferencing will be handled. I have spent a lot of time looking for definitive information on the subject with no luck. Well, maybe too much luck. There seem to be many proposals and models and so on, but it is not clear to me what is actually being done in practice. I have downloaded all the RFCs and proposal papers on the subject. I am still reviewing them, but I think the folks on this forum can help me a lot.
I need to know the SIP message sequences for performing a call transfer and a blind call transfer and for constructing a conference. I have found information in proposals, but I need to know what actual, available SIP phones can do. We have some phones that we will test, but I do not know what they do when you press their transfer and conference buttons. Pardon me again for my impatience in asking before I have tried this out.
The Transfer models are straightforward, but conferencing is more complicated. We must construct a simple conferencing model where the conferencing is performed by a central server, a SIP IPX. Only conferences of 3 participants need to be supported. We want it to look exactly like 3-way calling on your home phone. During a call, put the call on hold with a conference button, call another phone, hit conference button, the two calls are joined in a 3-way conference.
The document draft-ietf-sipping-service-examples-07.txt seems to be very helpful on the subject, but all examples are in the form of 3 or more UAs and do not address any examples from the point of view of a PBX. I can see how to extend the examples to a PBX case, except for one aspect. If the IP-PBX is to perform the action as a proxy, what does the phone send the IP-PBX to indicate the steps in the process. Put more plainly, what happens when the user hits the Transfer or Conference button on the phone? What message is sent to the IP-PBX?
Can anyone tell me where else I should be looking? Is the service examples draft the best base document to work from?
Thanks in advance, Bob Carlson
Bob,
I can offer some ideas that might help. I certainly don't intend to be condescending...
Many of my questions are answered with call traces. For example, I worked on a bug with REINVITES today.
http://www.addaline.com/traces/andy_index.html
This is created by using a :
1) switch with port monitoring 2) ethereal (or tcpdump) to grab data 3) sipscenario to format the data into the call trace
A transfer can be done in a few different ways, especially when you get an IP-PBX involved. There is a popular one called Asterisk that can do transfers between extensions. If you built it, get phones to register with it, and connected the outside with a SIP provider you could do some call traces and see how Asterisk makes it happen.
A conference is a different animal. I don't think there is any SIP call per se to build a conference. Some UAs have the function built in, and they actually create more than one phone call and mix the sound internally. For example, the Cisco 7960 IP phone does that.
I guess the basic problem is that SIP is a protocol, transfer is a feature that is implemented with the SIP protocol. There are quite a few ways to skin that cat :-)
-g
On Nov 4, 2004, at 4:46 PM, Bob Carlson wrote:
I sent this earlier and got no responses. Perhaps this is not the right forum to ask this question. Can any one suggest a better place to go for this information?
Thanks, Bob
-----Original Message----- From: Bob Carlson Sent: Wednesday, November 03, 2004 3:22 PM To: 'SerUsers' Subject: Transfer and Conferencing
Let me apologize in advance for my question, which is a little rudimentary. We are just starting a project that will use SER and I am being forced to document right now how transfer and conferencing will be handled. I have spent a lot of time looking for definitive information on the subject with no luck. Well, maybe too much luck. There seem to be many proposals and models and so on, but it is not clear to me what is actually being done in practice. I have downloaded all the RFCs and proposal papers on the subject. I am still reviewing them, but I think the folks on this forum can help me a lot.
I need to know the SIP message sequences for performing a call transfer and a blind call transfer and for constructing a conference. I have found information in proposals, but I need to know what actual, available SIP phones can do. We have some phones that we will test, but I do not know what they do when you press their transfer and conference buttons. Pardon me again for my impatience in asking before I have tried this out.
The Transfer models are straightforward, but conferencing is more complicated. We must construct a simple conferencing model where the conferencing is performed by a central server, a SIP IPX. Only conferences of 3 participants need to be supported. We want it to look exactly like 3-way calling on your home phone. During a call, put the call on hold with a conference button, call another phone, hit conference button, the two calls are joined in a 3-way conference.
The document draft-ietf-sipping-service-examples-07.txt seems to be very helpful on the subject, but all examples are in the form of 3 or more UAs and do not address any examples from the point of view of a PBX. I can see how to extend the examples to a PBX case, except for one aspect. If the IP-PBX is to perform the action as a proxy, what does the phone send the IP-PBX to indicate the steps in the process. Put more plainly, what happens when the user hits the Transfer or Conference button on the phone? What message is sent to the IP-PBX?
Can anyone tell me where else I should be looking? Is the service examples draft the best base document to work from?
Thanks in advance, Bob Carlson
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Greg Fausak www.AddaBrand.com (US) 469-546-1265
Thanks a lot for the pointer to Sip Scenario. It looks like a fantastic tool. I will take a look at Asterisk.
Thanks again, Bob
-----Original Message----- From: Greg Fausak [mailto:greg@addabrand.com] Sent: Thursday, November 04, 2004 3:23 PM To: Bob Carlson Cc: 'SerUsers' Subject: Re: [Serusers] RE: Transfer and Conferencing - Please help!
Bob,
I can offer some ideas that might help. I certainly don't intend to be condescending...
Many of my questions are answered with call traces. For example, I worked on a bug with REINVITES today.
http://www.addaline.com/traces/andy_index.html
This is created by using a :
1) switch with port monitoring 2) ethereal (or tcpdump) to grab data 3) sipscenario to format the data into the call trace
A transfer can be done in a few different ways, especially when you get an IP-PBX involved. There is a popular one called Asterisk that can do transfers between extensions. If you built it, get phones to register with it, and connected the outside with a SIP provider you could do some call traces and see how Asterisk makes it happen.
A conference is a different animal. I don't think there is any SIP call per se to build a conference. Some UAs have the function built in, and they actually create more than one phone call and mix the sound internally. For example, the Cisco 7960 IP phone does that.
I guess the basic problem is that SIP is a protocol, transfer is a feature that is implemented with the SIP protocol. There are quite a few ways to skin that cat :-)
-g
On Nov 4, 2004, at 4:46 PM, Bob Carlson wrote:
I sent this earlier and got no responses. Perhaps this is not the right forum to ask this question. Can any one suggest a better place to go for this information?
Thanks, Bob
-----Original Message----- From: Bob Carlson Sent: Wednesday, November 03, 2004 3:22 PM To: 'SerUsers' Subject: Transfer and Conferencing
Let me apologize in advance for my question, which is a little rudimentary. We are just starting a project that will use SER and I am being forced to document right now how transfer and conferencing will be handled. I have spent a lot of time looking for definitive information on the subject with no luck. Well, maybe too much luck. There seem to be many proposals and models and so on, but it is not clear to me what is actually being done in practice. I have downloaded all the RFCs and proposal papers on the subject. I am still reviewing them, but I think the folks on this forum can help me a lot.
I need to know the SIP message sequences for performing a call transfer and a blind call transfer and for constructing a conference. I have found information in proposals, but I need to know what actual, available SIP phones can do. We have some phones that we will test, but I do not know what they do when you press their transfer and conference buttons. Pardon me again for my impatience in asking before I have tried this out.
The Transfer models are straightforward, but conferencing is more complicated. We must construct a simple conferencing model where the conferencing is performed by a central server, a SIP IPX. Only conferences of 3 participants need to be supported. We want it to look exactly like 3-way calling on your home phone. During a call, put the call on hold with a conference button, call another phone, hit conference button, the two calls are joined in a 3-way conference.
The document draft-ietf-sipping-service-examples-07.txt seems to be very helpful on the subject, but all examples are in the form of 3 or more UAs and do not address any examples from the point of view of a PBX. I can see how to extend the examples to a PBX case, except for one aspect. If the IP-PBX is to perform the action as a proxy, what does the phone send the IP-PBX to indicate the steps in the process. Put more plainly, what happens when the user hits the Transfer or Conference button on the phone? What message is sent to the IP-PBX?
Can anyone tell me where else I should be looking? Is the service examples draft the best base document to work from?
Thanks in advance, Bob Carlson
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Greg Fausak www.AddaBrand.com (US) 469-546-1265
.. this is proberbly not what you want, but look into sems, sems has a confrancing module :-) And it works Gr8
-Atle
* Bob Carlson bob.carlson@sigpro.com [041105 00:46]:
Thanks a lot for the pointer to Sip Scenario. It looks like a fantastic tool. I will take a look at Asterisk.
Thanks again, Bob
-----Original Message----- From: Greg Fausak [mailto:greg@addabrand.com] Sent: Thursday, November 04, 2004 3:23 PM To: Bob Carlson Cc: 'SerUsers' Subject: Re: [Serusers] RE: Transfer and Conferencing - Please help!
Bob,
I can offer some ideas that might help. I certainly don't intend to be condescending...
Many of my questions are answered with call traces. For example, I worked on a bug with REINVITES today.
http://www.addaline.com/traces/andy_index.html
This is created by using a :
- switch with port monitoring
- ethereal (or tcpdump) to grab data
- sipscenario to format the data into the call trace
A transfer can be done in a few different ways, especially when you get an IP-PBX involved. There is a popular one called Asterisk that can do transfers between extensions. If you built it, get phones to register with it, and connected the outside with a SIP provider you could do some call traces and see how Asterisk makes it happen.
A conference is a different animal. I don't think there is any SIP call per se to build a conference. Some UAs have the function built in, and they actually create more than one phone call and mix the sound internally. For example, the Cisco 7960 IP phone does that.
I guess the basic problem is that SIP is a protocol, transfer is a feature that is implemented with the SIP protocol. There are quite a few ways to skin that cat :-)
-g
On Nov 4, 2004, at 4:46 PM, Bob Carlson wrote:
I sent this earlier and got no responses. Perhaps this is not the right forum to ask this question. Can any one suggest a better place to go for this information?
Thanks, Bob
-----Original Message----- From: Bob Carlson Sent: Wednesday, November 03, 2004 3:22 PM To: 'SerUsers' Subject: Transfer and Conferencing
Let me apologize in advance for my question, which is a little rudimentary. We are just starting a project that will use SER and I am being forced to document right now how transfer and conferencing will be handled. I have spent a lot of time looking for definitive information on the subject with no luck. Well, maybe too much luck. There seem to be many proposals and models and so on, but it is not clear to me what is actually being done in practice. I have downloaded all the RFCs and proposal papers on the subject. I am still reviewing them, but I think the folks on this forum can help me a lot.
I need to know the SIP message sequences for performing a call transfer and a blind call transfer and for constructing a conference. I have found information in proposals, but I need to know what actual, available SIP phones can do. We have some phones that we will test, but I do not know what they do when you press their transfer and conference buttons. Pardon me again for my impatience in asking before I have tried this out.
The Transfer models are straightforward, but conferencing is more complicated. We must construct a simple conferencing model where the conferencing is performed by a central server, a SIP IPX. Only conferences of 3 participants need to be supported. We want it to look exactly like 3-way calling on your home phone. During a call, put the call on hold with a conference button, call another phone, hit conference button, the two calls are joined in a 3-way conference.
The document draft-ietf-sipping-service-examples-07.txt seems to be very helpful on the subject, but all examples are in the form of 3 or more UAs and do not address any examples from the point of view of a PBX. I can see how to extend the examples to a PBX case, except for one aspect. If the IP-PBX is to perform the action as a proxy, what does the phone send the IP-PBX to indicate the steps in the process. Put more plainly, what happens when the user hits the Transfer or Conference button on the phone? What message is sent to the IP-PBX?
Can anyone tell me where else I should be looking? Is the service examples draft the best base document to work from?
Thanks in advance, Bob Carlson
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Greg Fausak www.AddaBrand.com (US) 469-546-1265
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Atle Samuelsen -
Bob Carlson -
Greg Fausak