[SR-Users] No Audio For Outbound Calls
Kashish Raheja
kashishraheja1809 at gmail.com
Thu May 20 22:14:57 CEST 2021
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 at 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 at 192.168.0.192:5060>;tag=as69eb1cce
+0.050579 │ *──────────────────────────>* │
│ │ │To:
<sip:09413745250 at 192.168.0.192:5060>
01:22:15.832728 │ 100 trying -- your call is │
│ │ │Contact:
<sip:68983619 at 3.236.72.101:5060>
+0.000348 │ <────────────────────────── │
│ │ │Call-ID:
1191aedf331ec3e35955bf376a20999d at 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 at 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 at 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 at 192.168.0.192:5060>;tag=as69eb1cce
+0.000348 │ <────────────────────────── │
│ │ │To:
<sip:09413745250 at 192.168.0.192:5060>
01:22:15.833076 │ │
│ * INVITE (SDP) * │ │Contact:
<sip:68983619 at 3.236.72.101:5060>
+0.004863 │ │
│ *──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d at 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 at 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 at 14.98.22.110
+0.050579 │ ──────────────────────────> │
│ │ │From:
<sip:68983619 at 192.168.0.192:5060>;tag=as69eb1cce
01:22:15.832728 │ 100 trying -- your call is │
│ │ │To:
<sip:09413745250 at 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 at 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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