[Freeswitch-dev] Session Progress and RINGING

Thu Nov 4 15:17:38 PDT 2010

try to patch it i guess.
I still say the RFC is to vague to call this behavior complying with the RFC.
if you can make a clean patch that we can wrap in a pram like
support-one-of-many-interpretation-of-rfc3960=true

On Thu, Nov 4, 2010 at 5:05 PM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> 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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>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> 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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>> >> >>
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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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>> >> FreeSWITCH-dev at lists.freeswitch.org
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>> >>
>> >
>> > _______________________________________________
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>> > FreeSWITCH-dev at lists.freeswitch.org
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>> > 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
>>
>> _______________________________________________
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>>
>
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-- 
Anthony Minessale II

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