[Freeswitch-users] Push to talk

[Bob Hartwig]

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 at 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 at 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 at 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 at 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 at 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
>>>>>
>>>>>
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>>>>
>>>>
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>>
>>
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>
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