Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-21 Thread Iñaki Baz Castillo
2010/3/21 Olle E. Johansson : > Fully agree. Why I keep being persistent here is that too many applications > assume that 183 means ringing. They just don't understand the importance of > the > state change of the call in the network - "the user will hear the ringing". > Audio > is not the only w

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-21 Thread Olle E. Johansson
20 mar 2010 kl. 19.45 skrev Pranab Bohra: > In my opinion, the decision on whether to respond with 183 or 180, > with or without SDP and other things related to early media should be > taken on case-by-case basis. > In case of analog termination, as in the present case, ring-tone or > progress in

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Paul Kyzivat
I agree that it is better to return a 180 if you know that alerting is happening. Unfortunately, the UAS doesn't always know that. Gateways in particular may not know. So the UAC has to be prepared to cope if the 180 doesn't arrive. Thanks, Paul Olle E. Johansson wrote: > 20 ma

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Pranab Bohra
In my opinion, the decision on whether to respond with 183 or 180, with or without SDP and other things related to early media should be taken on case-by-case basis. In case of analog termination, as in the present case, ring-tone or progress indication tone is always inband. Hence the UAC should b

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Olle E. Johansson
20 mar 2010 kl. 16.40 skrev Iñaki Baz Castillo: > 2010/3/20 Olle E. Johansson : > >>> For example, some TDM nodes fake the ringing and generate real audio. >>> When such TDM signaling arrives to a SIP-PSTN gateway, it must >>> generate a 183 with SDP :( > >> NO, IT can generate 180 with SDP as

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Paul Kyzivat
Olle E. Johansson wrote: > 20 mar 2010 kl. 15.50 skrev Iñaki Baz Castillo: > >> 2010/3/20 Pranab Bohra : >>> As per section 2.1 of rfc 3666, the softphone should open the rtp port >>> to receive early media when the gateway responds with 183 + SDP. >> The problem is that some servers/gateways s

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Iñaki Baz Castillo
2010/3/20 Olle E. Johansson : >> For example, some TDM nodes fake the ringing and generate real audio. >> When such TDM signaling arrives to a SIP-PSTN gateway, it must >> generate a 183 with SDP :( > NO, IT can generate 180 with SDP as default... IMHO there is no difference at all between a 180

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Olle E. Johansson
20 mar 2010 kl. 15.50 skrev Iñaki Baz Castillo: > 2010/3/20 Pranab Bohra : >> As per section 2.1 of rfc 3666, the softphone should open the rtp port >> to receive early media when the gateway responds with 183 + SDP. > > The problem is that some servers/gateways send first a 183 with SDP. > The

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Olle E. Johansson
20 mar 2010 kl. 14.52 skrev Iñaki Baz Castillo: > 2010/3/20 Olle E. Johansson : >>> Also, there are cases in which the PSTN gateways reply a 183 followed >>> by a 180. This makes crazy some phones which don't know what to render >>> (real audio of 183 or artificial ringing requested by the 180).

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Iñaki Baz Castillo
2010/3/20 Paul Kyzivat : >> The UAC receives the early media. But after a second (or less) the >> server sends a 180 and *stops* the RTP. >> If the UAC chooses to render the real audio (183) rather than the >> artificial ringing (180) then there will be no audio anymore (until >> the 200 arrive of

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Paul Kyzivat
Iñaki Baz Castillo wrote: > 2010/3/20 Pranab Bohra : >> As per section 2.1 of rfc 3666, the softphone should open the rtp port >> to receive early media when the gateway responds with 183 + SDP. > > The problem is that some servers/gateways send first a 183 with SDP. > The UAC receives the earl

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Iñaki Baz Castillo
2010/3/20 Pranab Bohra : > As per section 2.1 of rfc 3666, the softphone should open the rtp port > to receive early media when the gateway responds with 183 +  SDP. The problem is that some servers/gateways send first a 183 with SDP. The UAC receives the early media. But after a second (or less)

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Pranab Bohra
As per section 2.1 of rfc 3666, the softphone should open the rtp port to receive early media when the gateway responds with 183 + SDP. On Sat, Mar 20, 2010 at 6:52 PM, Olle E. Johansson wrote: > > 20 mar 2010 kl. 13.43 skrev Iñaki Baz Castillo: > >> 2010/3/20 Olle E. Johansson : >>> >>> 18 mar

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Iñaki Baz Castillo
2010/3/20 Olle E. Johansson : >> Also, there are cases in which the PSTN gateways reply a 183 followed >> by a 180. This makes crazy some phones which don't know what to render >> (real audio of 183 or artificial ringing requested by the 180). >> > In the case of using early media just to send a ri

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Olle E. Johansson
20 mar 2010 kl. 13.43 skrev Iñaki Baz Castillo: > 2010/3/20 Olle E. Johansson : >> >> 18 mar 2010 kl. 12.34 skrev Iñaki Baz Castillo: >>> Nothing special, the UAC should b able to receive such RTP (real >>> audio) and render it to the human via the phone speaker. >>> It's really common and 200%

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Iñaki Baz Castillo
2010/3/20 Olle E. Johansson : > > 18 mar 2010 kl. 12.34 skrev Iñaki Baz Castillo: >> Nothing special, the UAC should b able to receive such RTP (real >> audio) and render it to the human via the phone speaker. >> It's really common and 200% valid. > > As a best current practise, I would recommend f

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-20 Thread Olle E. Johansson
18 mar 2010 kl. 12.34 skrev Iñaki Baz Castillo: > 2010/3/18 Rodriguez Merchan, Pedro Julian : >> What must i do when i receive the SESSION PROGRESS instead the RINGING? > > Nothing special, the UAC should b able to receive such RTP (real > audio) and render it to the human via the phone speaker.

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-18 Thread Kamalakanta Palei (kpalei)
Message- From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Rodriguez Merchan, Pedro Julian Sent: Thursday, March 18, 2010 4:51 PM To: Sip-implementors@lists.cs.columbia.edu Subject: [Sip-implementors] Managing sip Session

Re: [Sip-implementors] Managing sip Session progress 183.

2010-03-18 Thread Iñaki Baz Castillo
2010/3/18 Rodriguez Merchan, Pedro Julian : > What must i do when i receive the SESSION PROGRESS instead the RINGING? Nothing special, the UAC should b able to receive such RTP (real audio) and render it to the human via the phone speaker. It's really common and 200% valid. -- Iñaki Baz Castillo

[Sip-implementors] Managing sip Session progress 183.

2010-03-18 Thread Rodriguez Merchan, Pedro Julian
Hi, Im having some problems managing a softphone when receive a session Progress indication instead a Ringing message. My enviroment is : AnalogFXO---FXS---Asterisk---SIP-my softphone. Phone + TDM ca