asterisk users - Dec 2018 - Outbound call: caller gets no ringback on session progress

Hello!

An extension registered at asterisk 13.23 initiates an external call (pjsip).
After the Invite, the
callee (-> ISP) sends a

	100 Trying
	183 Session Progress (*without* SDP)


Asterisk now sends to the extension:

	183 Session Progress (*with* SDP)
	183 Session Progress (*with* SDP) (really two times)
	

The callee meanwhile sends

	180 Ringing (*without* SDP)

which is "forwarded" by Asterisk to the extension with

	180 Ringing (*with* SDP)


The problem: The extension doesn't create a ringback locally, because it
most probably expects it to
be sent by the callee - but the callee doesn't send anything (not
surprising, because there has been
no SDP).

Or should Asterisk create the ringback (Asterisk doesn't send any RTP
package)? Or should the phone
create the ringback itself because there is a 180 Ringing (even if it contains
SDP)?

I'm wondering: Why does Asterisk create a 183 to the extension containing
SDP if the callee didn't
provide any SDP?


So many questions ... . Could somebody please shine some light on it? What's
going wrong here?



Thanks,
Michael

On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:

<snip>
> 
> The problem: The extension doesn't create a ringback locally, because 
> it most probably expects it to
> be sent by the callee - but the callee doesn't send anything (not 
> surprising, because there has been
> no SDP).
> 
> Or should Asterisk create the ringback (Asterisk doesn't send any RTP 
> package)? Or should the phone
> create the ringback itself because there is a 180 Ringing (even if it 
> contains SDP)?
> 
> I'm wondering: Why does Asterisk create a 183 to the extension 
> containing SDP if the callee didn't
> provide any SDP?
> 
> 
> So many questions ... . Could somebody please shine some light on it? 
> What's going wrong here?
The core doesn't communicate whether progress includes media or not, so the
PJSIP channel driver (and even chan_sip) assumes media is there. What should
happen in chan_pjsip is that it would send inband ringing and not a 180 Ringing
with SDP, but that is not currently implemented and I don't think this
particularly interaction has come up before to cause it to be implemented.
I'd therefore suggest raising an issue[1] with the SIP trace.

[1] https://issues.asterisk.org/jira

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

On 12.12.18 at 19:43 Joshua C. Colp wrote:
> On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:
> 
> <snip>
> 
>>
>> The problem: The extension doesn't create a ringback locally,
because
>> it most probably expects it to
>> be sent by the callee - but the callee doesn't send anything (not 
>> surprising, because there has been
>> no SDP).
>>
>> Or should Asterisk create the ringback (Asterisk doesn't send any
RTP
>> package)? Or should the phone
>> create the ringback itself because there is a 180 Ringing (even if it 
>> contains SDP)?
>>
>> I'm wondering: Why does Asterisk create a 183 to the extension 
>> containing SDP if the callee didn't
>> provide any SDP?
>>
>>
>> So many questions ... . Could somebody please shine some light on it? 
>> What's going wrong here?
> 
> The core doesn't communicate whether progress includes media or not, so
the PJSIP channel driver (and even chan_sip) assumes media is there. What should
happen in chan_pjsip is that it would send inband ringing and not a 180 Ringing
with SDP, but that is not currently implemented and I don't think this
particularly interaction has come up before to cause it to be implemented.
I'd therefore suggest raising an issue[1] with the SIP trace.
https://issues.asterisk.org/jira/browse/ASTERISK-28208


Thanks,
Michael

On 12.12.18 at 19:43 Joshua C. Colp wrote:
> On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:
> 
> <snip>
> 
>>
>> The problem: The extension doesn't create a ringback locally,
because
>> it most probably expects it to
>> be sent by the callee - but the callee doesn't send anything (not 
>> surprising, because there has been
>> no SDP).
>>
>> Or should Asterisk create the ringback (Asterisk doesn't send any
RTP
>> package)? Or should the phone
>> create the ringback itself because there is a 180 Ringing (even if it 
>> contains SDP)?
>>
>> I'm wondering: Why does Asterisk create a 183 to the extension 
>> containing SDP if the callee didn't
>> provide any SDP?
>>
>>
>> So many questions ... . Could somebody please shine some light on it? 
>> What's going wrong here?
> 
> The core doesn't communicate whether progress includes media or not, so
the PJSIP channel driver (and even chan_sip) assumes media is there.
Another question: is there any use case for 183 Session Progress w/o SDP? IOW:
Is a 183 Session
Progress just a bug of the ISP? If so, problem could be solved by dropping each
183 package w/o SDP.


Thanks,
Michael