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
Iñaki Baz Castillo wrote:
> 2010/3/20 Schwarz Albrecht :
>> It's amazing that such a fundamental session configuration topic is not
>> really well specified (explicitly, detailed, unambiguous) in RFCs 3261 &
>> 3264. Just wondering ...
>
> I agree. When such kind of issues occur it means that
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
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
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
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
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
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).
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
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
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)
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
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
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%
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
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.
The SIP Forum and the IPv6 Forum just announced a new partnership, to promote
interoperability and help the members and the industry to move forward with
IPv6. For all of us working with SIP, this is an important message.
I think we should try to add some automated self tests for IPv6 to the SIP
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