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
SDP is probably wrong, a trace here would help.
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Mon, May 10, 2021 at 10:11 AM 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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions
- sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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
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@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
Kamailio - Users Mailing List - Non Commercial Discussions
- sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
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
So let me get this right:
asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco op
There's something i'm not seeing. Can you explain further like i did above?
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Mon, May 10, 2021 at 8:42 PM Kashish Raheja 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:
[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
Kamailio - Users Mailing List - Non Commercial Discussions
- sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Hello,
Could you share your Asterisk's sip context where you set up connectivity with telco? Just delete all usernames and passwords, if any....
Your setup is very similar to the one I have with a very large telco in my country, and it works fine as long as you always keep Asterisk on the RTP path (no re-invites allowed, NAT always active)
*Sérgio Charrua*
*www.voip.pt http://www.voip.pt/* Tel.: +351 callto:+351+91+104+12+6621 130 71 77
Email : *sergio.charrua@voip.pt sergio.charrua@voip.pt*
This message and any files or documents attached are strictly confidential or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the named addressee or have received this email in error, please inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free.
On Tue, May 11, 2021 at 8:02 AM David Villasmil < david.villasmil.work@gmail.com> wrote:
So let me get this right:
asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco op
There's something i'm not seeing. Can you explain further like i did above?
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Mon, May 10, 2021 at 8:42 PM Kashish Raheja < 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:
[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
Kamailio - Users Mailing List - Non Commercial Discussions
- sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
Kamailio - Users Mailing List - Non Commercial Discussions
- sr-users@lists.kamailio.org
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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 actual flow: (Asterisk) (Kamailio Local ILL Port) (Kamailio Local SIP Interface) (Telecom Operator SBC)
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 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 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:
[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
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 192.168.0.192:5060 10.0.87.230:5060 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 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> 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
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
- sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
You need to also
INVITE
Asterisk->kamailio offer the asterisk’s public IP in the SDP. Kamailio->telco offer in the invite the rtpproxy’s prívate IP (10.x.x.x)
And on the 200
Kamailio->asterisk offer the rtpproxy’s PUBLIC ip.
On the cloud firewall, you need to allow the rtpproxy and Kamailio’s public IPs. On the asterisk, you need to configure its public IP as the SDP, not the private.
You also need to start rtpproxy in bridge mode providing both the public and private IPs at start (or config)
Hope this helps.
On Tue, 11 May 2021 at 09:32, Daniel Donoghue daniel.donoghue@freespee.com wrote:
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.
IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom operator carrier through this port.
- One of them is connected to the SIP trunk and this NIC port local
having local IP as 192.168.0.192 and public IP as 14.X.X.X
- Second NIC port is connected to ILL for internet connection
- 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 192.168.0.192:5060 10.0.87.230:5060 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> 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> 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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Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 70 01:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: sip:68983619@3.236.72.101:5060 +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE +0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0 01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer +0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263 01:22:18.127775 │ │ │ 180 Ringing │ │ +0.000189 │ │ │ <────────────────────────── │ │v=0 01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.000206 │ │ │ <<<──────────────────────── │ │t=0 0 01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101 +19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16 +0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150 01:22:37.974479 │ ACK │ │ │ │a=sendrecv +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Record-Route: sip:192.168.0.192;lr=on;ftag=as69eb1cce ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0 01:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060 +0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69 01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: sip:68983619@3.236.72.101:5060 +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE +0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer +1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp +0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279 01:22:18.127964 │ 180 Ringing │ │ │ │ +0.349351 │ <────────────────────────── │ │ │ │v=0 01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +19.181387 │ <<<──────────────────────── │ │ │ │t=0 0 01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000 +0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000 01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16 +0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150 01:22:37.974761 │ │ │ ACK │ │a=sendrecv +4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: sip:192.168.0.192;lr;ftag=as69eb1cce 01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-t7ln3f58c3ea1 +0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE 01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE +0.004863 │ │ │ ──────────────────────────> │ │Contact: sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff 01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0 +0.799120 │ │ │ <────────────────────────── │ │Require: timer 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac +0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │ 01:22:18.127775 │ │ │ 180 Ringing │ │v=0 +0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0 +0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101 01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000 +19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1 +0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes 01:22:37.732627 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 01:22:37.974479 │ ACK │ │ │ │ +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 http://127.0.0.1:7722 -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 http://192.168.0.192/10.0.87.230*
Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for incoming calls
Anything am I missing here?
Thanks. Regards Kashish
Haven't been able to sort this out yet. Anything am I missing here?
Thanks. Regards Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja kashishraheja1809@gmail.com wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 7001:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: sip:68983619@3.236.72.101:5060 +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE +0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0 01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer +0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263 01:22:18.127775 │ │ │ 180 Ringing │ │ +0.000189 │ │ │ <────────────────────────── │ │v=0 01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.000206 │ │ │ <<<──────────────────────── │ │t=0 0 01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101 +19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16 +0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150 01:22:37.974479 │ ACK │ │ │ │a=sendrecv +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Record-Route: <sip:192.168.0.192;lr=on;ftag=as69eb1cce> ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.001:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060 +0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69 01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: sip:68983619@3.236.72.101:5060 +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE +0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer +1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp +0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279 01:22:18.127964 │ 180 Ringing │ │ │ │ +0.349351 │ <────────────────────────── │ │ │ │v=0 01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +19.181387 │ <<<──────────────────────── │ │ │ │t=0 0 01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000 +0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000 01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16 +0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150 01:22:37.974761 │ │ │ ACK │ │a=sendrecv +4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: <sip:192.168.0.192;lr;ftag=as69eb1cce>01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-t7ln3f58c3ea1 +0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE 01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE +0.004863 │ │ │ ──────────────────────────> │ │Contact: sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff 01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0 +0.799120 │ │ │ <────────────────────────── │ │Require: timer 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac +0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │ 01:22:18.127775 │ │ │ 180 Ringing │ │v=0 +0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0 +0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101 01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000 +19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1 +0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes 01:22:37.732627 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 01:22:37.974479 │ ACK │ │ │ │ +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 http://127.0.0.1:7722 -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 http://192.168.0.192/10.0.87.230*
Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for incoming calls
Anything am I missing here?
Thanks. Regards Kashish
Kashish,
Best you can do is take a trace on BOTH ends and share it with us.
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Thu, May 27, 2021 at 2:26 PM Kashish Raheja kashishraheja1809@gmail.com wrote:
Haven't been able to sort this out yet. Anything am I missing here?
Thanks. Regards Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja < kashishraheja1809@gmail.com> wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 7001:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: sip:68983619@3.236.72.101:5060 +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE +0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0 01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer +0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263 01:22:18.127775 │ │ │ 180 Ringing │ │ +0.000189 │ │ │ <────────────────────────── │ │v=0 01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.000206 │ │ │ <<<──────────────────────── │ │t=0 0 01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101 +19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16 +0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150 01:22:37.974479 │ ACK │ │ │ │a=sendrecv +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Record-Route: <sip:192.168.0.192;lr=on;ftag=as69eb1cce> ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.001:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060 +0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69 01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: sip:68983619@3.236.72.101:5060 +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE +0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer +1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp +0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279 01:22:18.127964 │ 180 Ringing │ │ │ │ +0.349351 │ <────────────────────────── │ │ │ │v=0 01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +19.181387 │ <<<──────────────────────── │ │ │ │t=0 0 01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000 +0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000 01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16 +0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150 01:22:37.974761 │ │ │ ACK │ │a=sendrecv +4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: <sip:192.168.0.192;lr;ftag=as69eb1cce>01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-t7ln3f58c3ea1 +0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE 01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE +0.004863 │ │ │ ──────────────────────────> │ │Contact: sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff 01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0 +0.799120 │ │ │ <────────────────────────── │ │Require: timer 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac +0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │ 01:22:18.127775 │ │ │ 180 Ringing │ │v=0 +0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0 +0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101 01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000 +19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1 +0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes 01:22:37.732627 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 01:22:37.974479 │ ACK │ │ │ │ +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 http://127.0.0.1:7722 -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 http://192.168.0.192/10.0.87.230*
Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for incoming calls
Anything am I missing here?
Thanks. Regards Kashish
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Yes, have shared the complete traces in this thread itself.
Thanks. Regards Kashish
On Thu, May 27, 2021 at 6:44 PM Kashish Raheja kashishraheja1809@gmail.com wrote:
Haven't been able to sort this out yet. Anything am I missing here?
Thanks. Regards Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja < kashishraheja1809@gmail.com> wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 7001:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: sip:68983619@3.236.72.101:5060 +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE +0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0 01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer +0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263 01:22:18.127775 │ │ │ 180 Ringing │ │ +0.000189 │ │ │ <────────────────────────── │ │v=0 01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.000206 │ │ │ <<<──────────────────────── │ │t=0 0 01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101 +19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16 +0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150 01:22:37.974479 │ ACK │ │ │ │a=sendrecv +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Record-Route: <sip:192.168.0.192;lr=on;ftag=as69eb1cce> ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.001:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060 +0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69 01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: sip:68983619@3.236.72.101:5060 +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE +0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer +1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp +0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279 01:22:18.127964 │ 180 Ringing │ │ │ │ +0.349351 │ <────────────────────────── │ │ │ │v=0 01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +19.181387 │ <<<──────────────────────── │ │ │ │t=0 0 01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000 +0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000 01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16 +0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150 01:22:37.974761 │ │ │ ACK │ │a=sendrecv +4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: <sip:192.168.0.192;lr;ftag=as69eb1cce>01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-t7ln3f58c3ea1 +0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE 01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE +0.004863 │ │ │ ──────────────────────────> │ │Contact: sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff 01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0 +0.799120 │ │ │ <────────────────────────── │ │Require: timer 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac +0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │ 01:22:18.127775 │ │ │ 180 Ringing │ │v=0 +0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0 +0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101 01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000 +19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1 +0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes 01:22:37.732627 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 01:22:37.974479 │ ACK │ │ │ │ +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 http://127.0.0.1:7722 -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 http://192.168.0.192/10.0.87.230*
Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for incoming calls
Anything am I missing here?
Thanks. Regards Kashish
It’s very difficult to read, don’t paste an sngrep screenshot. If you can, please save the call to a file, then share that. :)
On Thu, 27 May 2021 at 21:09, Kashish Raheja kashishraheja1809@gmail.com wrote:
Yes, have shared the complete traces in this thread itself.
Thanks. Regards Kashish
On Thu, May 27, 2021 at 6:44 PM Kashish Raheja < kashishraheja1809@gmail.com> wrote:
Haven't been able to sort this out yet. Anything am I missing here?
Thanks. Regards Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja < kashishraheja1809@gmail.com> wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't seem to work. No audio in the outbound call however incoming call works fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;rport ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Max-Forwards: 7001:22:15.782149 │ *INVITE (SDP) * │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.050579 │ *──────────────────────────>* │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.832728 │ 100 trying -- your call is │ │ │ │Contact: sip:68983619@3.236.72.101:5060 +0.000348 │ <────────────────────────── │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.833076 │ │ │ INVITE (SDP) │ │CSeq: 102 INVITE +0.004863 │ │ │ ──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0 01:22:15.837939 │ │ │ 100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.799120 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Supported: replaces, timer +0.000179 │ │ │ <────────────────────────── │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │Content-Length: 263 01:22:18.127775 │ │ │ 180 Ringing │ │ +0.000189 │ │ │ <────────────────────────── │ │v=0 01:22:18.127964 │ 180 Ringing │ │ │ │o=root 1560151942 1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.349351 │ <────────────────────────── │ │ │ │s=Asterisk PBX 17.7.0 01:22:18.477315 │ │ │ 180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)* +0.000206 │ │ │ <<<──────────────────────── │ │t=0 0 01:22:18.477521 │ 180 Ringing │ │ │ │m=audio 14046 RTP/AVP 8 0 101 +19.181387 │ <<<──────────────────────── │ │ │ │a=rtpmap:8 PCMA/8000 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:101 telephone-event/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=fmtp:101 0-16 +0.241852 │ <────────────────────────── │ │ │ │a=maxptime:150 01:22:37.974479 │ ACK │ │ │ │a=sendrecv +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Record-Route: <sip:192.168.0.192;lr=on;ftag=as69eb1cce> ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.001:22:15.782149 │ INVITE (SDP) │ │ │ │Via: SIP/2.0/UDP 3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060 +0.050579 │ ──────────────────────────> │ │ │ │Max-Forwards: 69 01:22:15.832728 │ 100 trying -- your call is │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce +0.000348 │ <────────────────────────── │ │ │ │To: sip:09413745250@192.168.0.192:5060 01:22:15.833076 │ │ │ * INVITE (SDP) * │ │Contact: sip:68983619@3.236.72.101:5060 +0.004863 │ │ │ *──────────────────────────>* │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 01:22:15.837939 │ │ │ 100 Trying │ │CSeq: 102 INVITE +0.799120 │ │ │ <────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT +0.000179 │ │ │ <────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Supported: replaces, timer +1.490537 │ <────────────────────────── │ │ │ │P-Preferred-Identity: sip:68983600@10.0.76.9 01:22:18.127775 │ │ │ 180 Ringing │ │Content-Type: application/sdp +0.000189 │ │ │ <────────────────────────── │ │Content-Length: 279 01:22:18.127964 │ 180 Ringing │ │ │ │ +0.349351 │ <────────────────────────── │ │ │ │v=0 01:22:18.477315 │ │ │ 180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +0.000206 │ │ │ <<<──────────────────────── │ │s=Asterisk PBX 17.7.0 01:22:18.477521 │ 180 Ringing │ │ │ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)* +19.181387 │ <<<──────────────────────── │ │ │ │t=0 0 01:22:37.658908 │ │ │ 200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101 +0.073719 │ │ │ <────────────────────────── │ │a=rtpmap:8 PCMA/8000 01:22:37.732627 │ 200 OK (SDP) │ │ │ │a=rtpmap:0 PCMU/8000 +0.241852 │ <────────────────────────── │ │ │ │a=rtpmap:101 telephone-event/8000 01:22:37.974479 │ ACK │ │ │ │a=fmtp:101 0-16 +0.000282 │ ──────────────────────────> │ │ │ │a=maxptime:150 01:22:37.974761 │ │ │ ACK │ │a=sendrecv +4.095171 │ │ │ ──────────────────────────> │ │a=nortpproxy:yes 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK 3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────│Record-Route: <sip:192.168.0.192;lr;ftag=as69eb1cce>01:22:15.782149 │ INVITE (SDP) │ │ │ │Call-ID: 1191aedf331ec3e35955bf376a20999d@14.98.22.110 +0.050579 │ ──────────────────────────> │ │ │ │From: sip:68983619@192.168.0.192:5060;tag=as69eb1cce 01:22:15.832728 │ 100 trying -- your call is │ │ │ │To: sip:09413745250@192.168.0.192:5060;tag=aa2c806-t7ln3f58c3ea1 +0.000348 │ <────────────────────────── │ │ │ │CSeq: 102 INVITE 01:22:15.833076 │ │ │ INVITE (SDP) │ │Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE +0.004863 │ │ │ ──────────────────────────> │ │Contact: sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff 01:22:15.837939 │ │ │ 100 Trying │ │User-Agent: ZTE Softswitch/1.0.0 +0.799120 │ │ │ <────────────────────────── │ │Require: timer 01:22:16.637059 │ │ │ 183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac +0.000179 │ │ │ <────────────────────────── │ │Content-Length: 208 01:22:16.637238 │ 183 Session Progress (SDP) │ │ │ │Content-Type: application/sdp +1.490537 │ <────────────────────────── │ │ │ │ 01:22:18.127775 │ │ │ 180 Ringing │ │v=0 +0.000189 │ │ │ <────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.127964 │ 180 Ringing │ │ │ │s=SBC call +0.349351 │ <────────────────────────── │ │ │ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)* 01:22:18.477315 │ │ │ 180 Ringing │ │t=0 0 +0.000206 │ │ │ <<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101 01:22:18.477521 │ 180 Ringing │ │ │ │a=rtpmap:101 telephone-event/8000 +19.181387 │ <<<──────────────────────── │ │ │ │a=fmtp:101 0-15 01:22:37.658908 │ │ │ 200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1 +0.073719 │ │ │ <────────────────────────── │ │a=nortpproxy:yes 01:22:37.732627 │ * 200 OK (SDP)* │ │ │ │ +0.241852 │ *<──────────────────────────* │ │ │ │ 01:22:37.974479 │ ACK │ │ │ │ +0.000282 │ ──────────────────────────> │ │ │ │ 01:22:37.974761 │ │ │ ACK │ │ +4.095171 │ │ │ ──────────────────────────> │ │ 01:22:42.069932 │ │ │ BYE │ │ +0.000361 │ │ │ <────────────────────────── │ │ 01:22:42.070293 │ BYE │ │ │ │ +0.244125 │ <────────────────────────── │ │ │ │ 01:22:42.314418 │ 200 OK │ │ │ │ +0.000275 │ ──────────────────────────> │ │ │ │ 01:22:42.314693 │ │ │ 200 OK │ │ │ │ │ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command: */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 http://127.0.0.1:7722 -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230 http://192.168.0.192/10.0.87.230*
Apart from this, in the Asterisk console I can see that the RTP packets are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160) Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for incoming calls
Anything am I missing here?
Thanks. Regards Kashish
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