[Freeswitch-dev] Session Progress and RINGING

Thu Nov 4 15:05:38 PDT 2010

We will try that out but its not "100%" the functionality as the sbc does signalize. If I understand you correct, the 183 with SDP will be forwarded to A with 180 with SDP. But this does change the functionality to A (first of all on SIP/ISDN gateways).

Therefore I would prefer the "clean" way with RFC 3960.

Would you include a patch like "support-rfc3960" configuration option to sofia config which will ignore the CF_EARLY_MEDIA test?

----- Original Message -----
From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, 04 Nov 2010 22:54:34 +0100
Subject: Re: [Freeswitch-dev] Session Progress and RINGING

> how about what I said:
> {ignore_early_media=ring_ready}
> 
> 
> 
> On Thu, Nov 4, 2010 at 4:29 PM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
> > Hi,
> >
> > yes, they are not really sure but I think that the given RFC does specify
> this correctly (this does not mean, that all the SIP devices work like
> that). I could understand that a change will maybe result in big troubles.
> Perhaps a option for mod_sofia to "support-rfc3960" would be a good
> solution. If you want I will write the patch because it should be "really
> simple" to ignore the CF_EARLY_MEDIA if the option is set, or?
> >
> > We get the Session Progress with SDP and later the 180 Ringing from a
> session border controller (I am not 100% sure, but I think its a a
> Audiocodes).  The 180 Ringing has to be sent towards A because A is a ISDN
> gateway.
> >
> > Would do you prefer? Thanks for your investigation.
> >
> > 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 22:09:48 +0100
> > Subject: Re: [Freeswitch-dev] Session Progress and RINGING
> >
> >
> >> 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
> >> >> >
> >> >> >
> >> >> >
> >> >> > _______________________________________________
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> >> >> >
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> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> 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
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> >> >>
> >> >> FreeSWITCH Developer Conference
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> >> >>
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> >> >
> >> > _______________________________________________
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> >>
> >>
> >>
> >> --
> >> 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
> >>
> >> _______________________________________________
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> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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> >>
> >
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> 
> 
> 
> -- 
> 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
> 
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