10 May
2021
10 May
'21
4:53 p.m.
Is the telecom operator on a private network? The 200 OK SDP is asking the
telco to send the rtp to 10.0.X.X.
The 200 OK (Kamailio->telco) the sdp says:
c=IN IP4 10.0.X.X
That should be an IP the telco can reach.
You need to configure kamailio and RTPProxy to set an IP the telco can
actually reach. And probably do it on both the INVITE and the 200 OK.
On the initial invite, you should do the the same.
On Mon, 10 May 2021 at 11:39, Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when
I make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
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