[SR-Users] No Audio For Outbound Calls

[Kashish Raheja]

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 at 14.98.22.110
From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
To: <sip:09413745250 at 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 at 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 at 14.98.22.110
From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
To: <sip:09413745250 at 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 at 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 at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20210510/293efb55/attachment.htm>