[SR-Users] IMS early media
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"
mnc001.mcc001.3gppnetwork.org. 1D IN NAPTR 20 50 "s" "SIP+D2T"
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 ?
> 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
>> 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 ?
>> 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:
>>> 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 ?
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> --Mojtaba Esfandiari.S
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