[SR-Users] IMS early media

Mojtaba mespio at gmail.com
Mon Jan 28 11:26:36 CET 2019


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
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