is there a way to respond to a sip 100 trying?

[Matias Ellera]

Hello,

We have this use case: Web sends an invite to an unregistered device, then the back end responds with a sip 100 trying.

Looks like there is no event on RTCSession that fires when the back end just sends back a 100 trying.

The closest event to it looks to be 'progress' but it fires only when >100 <200 response comes back. 

Is there any way to tell when the back end returns just a sip 100 trying?

Thanks!

[Alex Balashov]

What are you actually trying to accomplish?

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Sent from mobile, with due apologies for brevity and errors.

On Sep 24, 2022, at 2:08 PM, Matias Ellera <ellera...@gmail.com> wrote:



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[Matt McGinnis]

We're looking to use the 100 trying as a trigger to play a ringback tone on the web client. The backend has users that are not connected to the PSTN and making calls between two end users.  We do not want to play a fake ringback in this case as we would like to have some level of backend server response to initiate the ringing and know the backend is working to setup the call.

Matias may weigh in with more info here too.

[Matt McGinnis]

To add, our use case does not easily allow a 180 ringing from the backend since these users are both serviced from the same backend and are not necessarily logged in and available at that time. There is no SIP phone registered and sitting there. When a user is not logged in, we rollover to a voicemail function.

[Alex Balashov]

It should be noted that 100 Trying isn’t considered a progress indication. It’s sent perfunctorily in response to virtually any invite, including one which will ultimately be rejected. Would you want to play ringback in such a situation? :)

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Sent from mobile, with due apologies for brevity and errors.

On Sep 24, 2022, at 3:30 PM, Matt McGinnis <mattmc...@gmail.com> wrote:

To add, our use case does not easily allow a 180 ringing from the backend since these users are both serviced from the same backend and are not necessarily logged in and available at that time. There is no SIP phone registered and sitting there. When a user is not logged in, we rollover to a voicemail function.

To view this discussion on the web visit https://groups.google.com/d/msgid/jssip/15a7d67e-07b1-45c0-8b2f-ac96c3f72ffbn%40googlegroups.com.