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:
[image: image.png]
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
