[SR-Users] IMS early media

Mojtaba mespio at gmail.com
Mon Jan 28 11:29:50 CET 2019


In another way, you could don't change this file, instead of change your
dial plan like below:
exten => 972551000002,1,Dial(
SIP/972551000002 at icscf.mnc001.mcc001.3gppnetwork.org,20
<SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20>);
WIth Regards.Mojtaba

On Mon, Jan 28, 2019 at 1:56 PM Mojtaba <mespio at gmail.com> wrote:

> It would be like these lines with afew changes:
> mnc001.mcc001.3gppnetwork.org.          1D IN A           10.82.10.56
> mnc001.mcc001.3gppnetwork.org.          1D IN NAPTR 10 50 "s"
> "SIP+D2U"    ""    _sip._udp
> mnc001.mcc001.3gppnetwork.org.          1D IN NAPTR 20 50 "s"
> "SIP+D2T"    ""    _sip._tcp
>
>
> On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev <
> tsvetan.filev at inno-networks.com> wrote:
>
>> Here is my current zone file:
>>
>> $ORIGIN mnc001.mcc001.3gppnetwork.org.
>> $TTL 1W
>> @                       1D IN SOA       localhost. root.localhost. (
>>                                         1               ; serial
>>                                         3H              ; refresh
>>                                         15M             ; retry
>>                                         1W              ; expiry
>>                                         1D )            ; minimum
>>
>>                         1D IN NS        ns
>> ns                      1D IN A         10.82.10.56
>>
>> pcscf                   1D IN A         10.82.10.56
>> _sip._udp.pcscf         1D SRV 0 0 5060 pcscf
>> _sip._tcp.pcscf         1D SRV 0 0 5060 pcscf
>>
>> icscf                   1D IN A         10.82.10.56
>> _sip._udp               1D SRV 0 0 4060 icscf
>> _sip._tcp               1D SRV 0 0 4060 icscf
>> _sip._udp.ims           1D SRV 0 0 4060 icscf
>> _sip._tcp.ims           1D SRV 0 0 4060 icscf
>>
>> scscf                   1D IN A         10.82.10.56
>> _sip._udp.scscf         1D SRV 0 0 6060 scscf
>> _sip._tcp.scscf         1D SRV 0 0 6060 scscf
>>
>> as                      1D IN A         10.82.10.56
>> _sip._udp.as            1D SRV 0 0 5062 as
>> _sip._tcp.as            1D SRV 0 0 5062 as
>>
>> hss                     1D IN A         10.82.10.56
>>
>>
>> How do I modify it in order to make this work ?
>>
>> Tnx.
>>
>> On 28.01.19 г. 11:50 ч., Mojtaba wrote:
>>
>> Hi Tsvetan,
>> Why do you send call back to S-CSCF? You should send call back to
>> I-CSCF.  Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
>> <SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20>, The ICSCF's IP
>> should be returned.
>>  Make sure entry SRV recordd in DNS server are true.
>> This kind of call back to IMS is true, But make sure you won't have any
>> issue in DNS resolve.
>>   exten => 972551000002,1,Dial(
>> SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20);
>>
>> With Regards.Mojtaba
>> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
>> tsvetan.filev at inno-networks.com> wrote:
>>
>>> 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> 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.
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
>>
>
> --
> --Mojtaba Esfandiari.S
>


-- 
--Mojtaba Esfandiari.S
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20190128/38b7b15c/attachment.html>


More information about the sr-users mailing list