Yes, the telecom operator is on the private network. 10.0.X.X is the SBC IP of the telecom operator to which we register. 10.0.X.X is reachable only through the second network interface. The complete flow is given below:

Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.

We are running the rtpproxy on local IP (192.168.0.192) in the following way:

rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s udp:localhost:7722

Thanks.

Regards

Kashish

On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <kashishraheja1809@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@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@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@gmail.com> wrote:

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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