[SR-Users] IMS early media

[Tsvetan Filev]

Hi Mojtaba.

I implemented the AS way and was able to play sound to the caller but In 
order to continue the call and send the invite to SCSCF I need to use 
proxy in the Dial application which is a problem (Asterisk is B2BUA not 
a proxy).
I found this old question here 
https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464 
that describes exactly the same issue.
Here is my dial plan:

exten => 972551000002,1,Progress()
exten => 972551000002,n,Playback(vm-starmain, noanswer)
exten => 972551000002,n,Wait(3)
exten => 972551000002,n,Hangup()
   ; This will send the call to the pcscf again
   ;  exten => 
972551000002,1,Dial(SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20);
   ; This will send the call to scscf but it will be rejected as domain 
not supported
   ;  exten => 
972551000002,1,Dial(SIP/972551000002 at scscf.mnc001.mcc001.3gppnetwork.org,20);

Can I use kamailio as an AS and implement the same ?

Regards.

On 22.12.18 г. 0:06 ч., Mojtaba wrote:
> Hello Tsvetan.
> Actually you could use SIP Early media in AS and also with cscf.
> If you choice the first way, i think it is very simple and 
> strightforward because you just use early media functions on your AS. 
> For example in Astrisk you could use Progress application and 'm' 
> option in Dial application in your dialplan.
> In second way you should check in Reply-Route block,if you got 180 
> ringing,  you have to use rtpproxy-stream funtion for doing sip early.
>
> Wih Regards.Mojtaba Esfandiari.S
>
> On Fri, 21 Dec 2018, 16:34 Tsvetan Filev, 
> <tsvetan.filev at inno-networks.com 
> <mailto:tsvetan.filev at inno-networks.com>> wrote:
>
>     Hi all.
>
>     I want to use SIP early media to play music to the caller in kamailio
>     IMS installation like this:
>     http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html
>
>     I looked a little bit but didn't find ready solution. The
>     information is
>     vague on this topic.
>     Should this be done through a module or application server ?
>     May I need to handle ringing in onreply_route and send OK with SDP to
>     the caller in SCSCF ?
>
>     Regards.
>
>
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