[SR-Users] No Audio For Outbound Calls
David Villasmil
david.villasmil.work at gmail.com
Fri May 28 10:29:08 CEST 2021
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 at 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 at 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 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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--
Regards,
David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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