Since your asterisk server is in the cloud, make sure the relevant udp audio port range is open to the telecoms carrier.
It is possible that your firewall/security group setup allows an incoming reply on a port when something first goes out on that port which can explain why audio works in one direction but not the other.
Blessings, — Daniel
On 11 May 2021, at 10:14, Kashish Raheja kashishraheja1809@gmail.com wrote:
This is how it looks:
Asterisk is running on Cloud having a public IP (3.236.X.X) Kamailio is running on a Physical server having 2 NIC ports. One of them is connected to the SIP trunk and this NIC port local IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom operator carrier through this port. Second NIC port is connected to ILL for internet connection having local IP as 192.168.0.192 and public IP as 14.X.X.X To make an outbound call, Asterisk Server (3.236.X.X) sends the call to Kamailio server on public IP (14.X.X.X) and in turn Kamailio server sends the call to telecom operator SBC (10.0.76.X) through 10.0.87.X port. Here is the diagram:
3.236.72.101:5060 <http://3.236.72.101:5060/> 192.168.0.192:5060 <http://192.168.0.192:5060/> 10.0.87.230:5060 <http://10.0.87.230:5060/> 10.0.76.9:5060 <http://10.0.76.9:5060/> ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬───────── 20:24:11.644416 │ INVITE (SDP) │ │ │ +0.000585 │ ──────────────────────────> │ │ │ 20:24:11.645001 │ 100 trying -- your call is │ │ │ +0.000235 │ <────────────────────────── │ │ │ 20:24:11.645236 │ │ │ INVITE (SDP) │ +0.005768 │ │ │ ──────────────────────────> │ 20:24:11.651004 │ │ │ 100 Trying │ +0.580627 │ │ │ <────────────────────────── │ 20:24:12.231631 │ │ │ 183 Session Progress (SDP) │ +0.000159 │ │ │ <────────────────────────── │ 20:24:12.231790 │ 183 Session Progress (SDP) │ │ │ +1.932655 │ <────────────────────────── │ │ │ 20:24:14.164445 │ │ │ 180 Ringing │ +0.000204 │ │ │ <────────────────────────── │ 20:24:14.164649 │ 180 Ringing │ │ │ +3.631157 │ <────────────────────────── │ │ │ 20:24:17.795806 │ │ │ 200 OK (SDP) │ +0.000361 │ │ │ <────────────────────────── │ 20:24:17.796167 │ 200 OK (SDP) │ │ │ +0.233102 │ <────────────────────────── │ │ │ 20:24:18.029269 │ ACK │ │ │ +0.000385 │ ──────────────────────────> │ │ │ 20:24:18.029654 │ │ │ ACK │ +11.647190 │ │ │ ──────────────────────────> │ 20:24:29.676844 │ │ │ BYE │ +0.000605 │ │ │ <────────────────────────── │ 20:24:29.677449 │ BYE │ │ │ +0.236993 │ <────────────────────────── │ │ │ 20:24:29.914442 │ 200 OK │ │ │ +0.000225 │ ──────────────────────────> │ │ │ 20:24:29.914667 │ │ │ 200 OK │ │ │ │ ──────────────────────────> │Thanks. Regards Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <kashishraheja1809@gmail.com mailto:kashishraheja1809@gmail.com> wrote: 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 mailto: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 mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110 From: sip:68XXXXX@10.0.X.X;tag=as2b21d944 To: <sip:09413745250@192.168.0.192:5060 http://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 http://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 mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110 From: sip:68XXXXX@10.0.X.X;tag=as2b21d944 To: <sip:09413745250@192.168.0.192:5060 http://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 mailto: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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions
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