[SR-Users] No Audio For Outbound Calls
Kashish Raheja
kashishraheja1809 at gmail.com
Thu May 27 15:14:36 CEST 2021
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 at 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 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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