[Freeswitch-dev] Session Progress and RINGING

[Anthony Minessale]

My conclusion from reading that discussion is they have no idea what to do.
There are votes both ways and 2 attempts to go of onto a tangent into
another topic.

We have chosen on our implementation to ignore 180 once we have
already established a media path from a 183.  The RFC is just vague
enough that this decision falls to us.

I understand that ISDN has more precise signaling in this regard.
PROGRESS or ALERTING both with or without media as a flag on the packet.

But the vast majority of SIP devices in the wild will ignore the 180
once early media has been established.
So even if we make changes to support this, it will be ignored by the
next guy in line.

Do you hear media during this early media phase?

This is why we have the originate param
{ignore_early_media=ring_ready} which will, in the case of sip,
translate 183 or 180 with and without SDP into a 180

To change it to do what you are asking for could have extreme negative
side effects and not even work in most devices so this is why I am
reluctant to change it.

Are you trying to cross connect 2 ISDN lines over SIP and preserve the
signalling?
If so, this is why SIP is flawed to begin with because it is lossy in
telephony signaling data.
This is why they now try to embed ss7 messages in the sip packets. =p



On Thu, Nov 4, 2010 at 3:51 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> session progress does have SDP which will then go through to A. But on the display of A it will only display "session progress" and not Ringing. If A is now a ISDN Gateway and has to signalize RINGING into the "ISDN world". I am not sure, if RINGING will be used for A to generate the ringback tone for itself on every device. This behaves on the client/device configuration.
>
> There was a nice discussion about this "issue" on the sip-implementors list:
>
> http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg06400.html
>
> There is a RFC for this case (Section: 3.2.  Ringing Tone Generation)
> http://www.rfc-editor.org/rfc/rfc3960.txt
>
> What do you think?
>
> Best regards,
> Bernhard Suttner
>
>
>
> ----- Original Message -----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Thu, 04 Nov 2010 16:54:30 +0100
> Subject: Re: [Freeswitch-dev] Session Progress and RINGING
>
>
>> no, once it gets session progress it will not send any ringing.
>> The sip side 180 ringing is used to tell the phone to generate its own
>> inband ringing.
>>
>> does your 183 session progress contain a sdp?
>>
>>
>> On Thu, Nov 4, 2010 at 10:25 AM, Bernhard Suttner
>> <bernhard.suttner at winet.ch> wrote:
>> > Hi,
>> >
>> > A ---> FS ---> B
>> >
>> > B does send a Session Progress to FS. FS will forward the Session Progress
>> to A.
>> > B does send a RINGING to FS. FS does _not_ forward this to A.
>> >
>> > Could it be, that the check the switch_test_flag(channel, CF_EARLY_MEDIA)
>> in mod_sofia.c within the INDICATE_RINGING section is wrong. I think it is
>> allowed to send the Session Progress first and later the RINGING. The
>> ringing is for example for SIP/ISDN gateways necessary.
>> >
>> > Any hint is appreciated.
>> >
>> > Best regards,
>> > Bernhard Suttner
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900