Is it possible to add push to talk capabilities to FreeSWITCH, perhaps via mod_conference?
its all software, anything is possible. That being said, the sip pieces to support PTT are not in place, so all that would need to be done, plus some facility for indicating PTT would have to be added, then support would need to be added in applications. Its doable but not a small amount of work.
On Dec 9, 2015, at 2:14 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
Is it possible to add push to talk capabilities to FreeSWITCH, perhaps via mod_conference?
Mike (and anyone else who reads this): Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation? I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT. Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM? Thanks, Larry Morley
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964 In theory, if a phone support the P-Answer-State header, it will work. On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Why not just use re-INVITEs with a=sendrecv and a=recvonly? On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Hi Bob Are you suggesting something along the lines of putting the callee's set - or sets belonging to members of s group - on hold after sending the initial INVITE? Or putting each group member's set on hold and leaving a call up - but on hold - until a caller leaves a group? So long as each set has two or more lines, normal calling should work, too: each user would have their regular extension and an extension that's dedicated to PTT type calls. Is that along the lines of what you're suggesting, or am I completely missing the mark? Thanks, Larry Why not just use re-INVITEs with a=sendrecv and a=recvonly? On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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PTT just means selectively muting and unmuting calls based on a switch closure or UI events, right? I know that some commercial conference systems implement muting by reinviting with a=sendonly, so if you're designing an end instrument that does PTT, it seems reasonable to implement that by reinviting with a=recvonly. Or if you're implementing the PTT functionality on the server side, use a=sendonly. Just spitballing here, I've never needed to do this. Bob On Sun, Dec 13, 2015 at 6:38 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Hi Bob
Are you suggesting something along the lines of putting the callee's set - or sets belonging to members of s group - on hold after sending the initial INVITE? Or putting each group member's set on hold and leaving a call up - but on hold - until a caller leaves a group?
So long as each set has two or more lines, normal calling should work, too: each user would have their regular extension and an extension that's dedicated to PTT type calls.
Is that along the lines of what you're suggesting, or am I completely missing the mark?
Thanks, Larry Why not just use re-INVITEs with a=sendrecv and a=recvonly?
On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley <lmorley@neny.cslimits.net
wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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It's more then that. You open a new channel, and instead of 180/3 you send 200, and then one side hears, while the other side talk. When done, there is an hangup, until some try to talk. On Dec 14, 2015 6:11 PM, "Bob Hartwig" <bobjectsfreeswitch@gmail.com> wrote:
PTT just means selectively muting and unmuting calls based on a switch closure or UI events, right? I know that some commercial conference systems implement muting by reinviting with a=sendonly, so if you're designing an end instrument that does PTT, it seems reasonable to implement that by reinviting with a=recvonly. Or if you're implementing the PTT functionality on the server side, use a=sendonly. Just spitballing here, I've never needed to do this.
Bob
On Sun, Dec 13, 2015 at 6:38 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Hi Bob
Are you suggesting something along the lines of putting the callee's set - or sets belonging to members of s group - on hold after sending the initial INVITE? Or putting each group member's set on hold and leaving a call up - but on hold - until a caller leaves a group?
So long as each set has two or more lines, normal calling should work, too: each user would have their regular extension and an extension that's dedicated to PTT type calls.
Is that along the lines of what you're suggesting, or am I completely missing the mark?
Thanks, Larry Why not just use re-INVITEs with a=sendrecv and a=recvonly?
On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley < lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Oh OK, completely disregard my suggestion then. I thought we were talking about a long-established multi-party conference with selective unmuting of end instruments. On Mon, Dec 14, 2015 at 10:37 AM, ik <idokan@gmail.com> wrote:
It's more then that. You open a new channel, and instead of 180/3 you send 200, and then one side hears, while the other side talk.
When done, there is an hangup, until some try to talk. On Dec 14, 2015 6:11 PM, "Bob Hartwig" <bobjectsfreeswitch@gmail.com> wrote:
PTT just means selectively muting and unmuting calls based on a switch closure or UI events, right? I know that some commercial conference systems implement muting by reinviting with a=sendonly, so if you're designing an end instrument that does PTT, it seems reasonable to implement that by reinviting with a=recvonly. Or if you're implementing the PTT functionality on the server side, use a=sendonly. Just spitballing here, I've never needed to do this.
Bob
On Sun, Dec 13, 2015 at 6:38 PM, Larry Morley <lmorley@neny.cslimits.net> wrote:
Hi Bob
Are you suggesting something along the lines of putting the callee's set - or sets belonging to members of s group - on hold after sending the initial INVITE? Or putting each group member's set on hold and leaving a call up - but on hold - until a caller leaves a group?
So long as each set has two or more lines, normal calling should work, too: each user would have their regular extension and an extension that's dedicated to PTT type calls.
Is that along the lines of what you're suggesting, or am I completely missing the mark?
Thanks, Larry Why not just use re-INVITEs with a=sendrecv and a=recvonly?
On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley < lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Bob, Thanks for taking the time to suggest that. While what you suggested might not be the magic bullet as far as a PTT system goes, it certainly seems to be the jumping off point for a different application I've been part contemplating and part messing with, on and off, for a while now. Thanks. Larry On Dec 14, 2015 12:00, "Bob Hartwig" <bobjectsfreeswitch@gmail.com> wrote:
Oh OK, completely disregard my suggestion then. I thought we were talking about a long-established multi-party conference with selective unmuting of end instruments.
On Mon, Dec 14, 2015 at 10:37 AM, ik <idokan@gmail.com> wrote:
It's more then that. You open a new channel, and instead of 180/3 you send 200, and then one side hears, while the other side talk.
When done, there is an hangup, until some try to talk. On Dec 14, 2015 6:11 PM, "Bob Hartwig" <bobjectsfreeswitch@gmail.com> wrote:
PTT just means selectively muting and unmuting calls based on a switch closure or UI events, right? I know that some commercial conference systems implement muting by reinviting with a=sendonly, so if you're designing an end instrument that does PTT, it seems reasonable to implement that by reinviting with a=recvonly. Or if you're implementing the PTT functionality on the server side, use a=sendonly. Just spitballing here, I've never needed to do this.
Bob
On Sun, Dec 13, 2015 at 6:38 PM, Larry Morley <lmorley@neny.cslimits.net
wrote:
Hi Bob
Are you suggesting something along the lines of putting the callee's set - or sets belonging to members of s group - on hold after sending the initial INVITE? Or putting each group member's set on hold and leaving a call up - but on hold - until a caller leaves a group?
So long as each set has two or more lines, normal calling should work, too: each user would have their regular extension and an extension that's dedicated to PTT type calls.
Is that along the lines of what you're suggesting, or am I completely missing the mark?
Thanks, Larry Why not just use re-INVITEs with a=sendrecv and a=recvonly?
On Sun, Dec 13, 2015 at 2:26 AM, ik <idokan@gmail.com> wrote:
The only PTT SIP protocol that I know of is the following: https://tools.ietf.org/html/rfc4964
In theory, if a phone support the P-Answer-State header, it will work.
On Fri, Dec 11, 2015 at 10:31 PM, Larry Morley < lmorley@neny.cslimits.net> wrote:
Mike (and anyone else who reads this):
Is there a standard for PTT on SIP, either formal or ad hoc? If not, do you know of anywhere there's a write up of an actual or a proposed implementation?
I'm interested in this functionality, too. We have a large public safety market here that makes heavy use of PTT.
Also, are you aware of any SIP capable PTT handsets? And any portables that can carry both, for example, 800 band radio and GSM?
Thanks, Larry Morley
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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participants (5)
-
Bob Hartwig
-
ik
-
Larry Morley
-
Michael Jerris
-
Michael Nielsen