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Dmitriy Serov

2016-Dec-16 20:19 UTC

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

Today I faced a problem. Please help to solve this problem.

Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware 
v2.06(AAGJ.9)C1

Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk).
Call using early media (183 Session in progress) and rtp_timeout=10.
After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] 
res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for lack
of RTP activity in 10 seconds

SIP dump is attached.

According to [1] before called user agent send OK or ACK there is one 
way SDP.
In sip dump (attached) i didn't find such SIP packets. Whether that call 
was canceled due to RTP inactivity?

Any help is welcome.

[1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt

-------------- next part --------------
INVITE sip:8xxx6yyy621 at txxx37.ru SIP/2.0
Via: SIP/2.0/UDP
11.111.11.11:5060;rport;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Max-Forwards: 70
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: sip:8xxx6yyy621 at txxx37.ru
Contact: "007" <sip:login at 11.111.11.11:5060;ob>
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
CSeq: 10072 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 90
Min-SE: 90
User-Agent: Keenetic Plus DECT
Authorization: Digest username="login", realm="ruvoip.net",
nonce="1481885583/bcb53e85a740689479f116a96fc7086b",
uri="sip:8xxx6yyy621 at txxx37.ru",
response="843f8211896b5b05fcf3a633d6d8eedf", algori
Content-Type: application/sdp
Content-Length:   326

v=0
o=- 3690874445 3690874445 IN IP4 11.111.11.11
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 0 8 109 96
c=IN IP4 11.111.11.11
b=TIAS:64000
a=rtcp:4023 IN IP4 11.111.11.11
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:109 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

[2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio TOS bits
184
[2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio CoS mark 5
[2016-12-16 13:53:03] VERBOSE[13985] res_pjsip_logger.c: <--- Transmitting
SIP response (352 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: <sip:8xxx6yyy621 at txxx37.ru>
CSeq: 10072 INVITE
Server: ruVoIP.net PBX
Content-Length:  0


[2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP
response (849 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: <sip:8xxx6yyy621 at
txxx37.ru>;tag=f30b688b-3358-4107-9992-fb6e3923bc15
CSeq: 10072 INVITE
Server: ruVoIP.net PBX
Contact: <sip:222.222.222.22:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, MESSAGE, REFER, REGISTER
Content-Type: application/sdp
Content-Length:   266

v=0
o=- 3690874445 3690874447 IN IP4 222.222.222.22
s=ruVoIP.net PBX
c=IN IP4 222.222.222.22
t=0 0
m=audio 25094 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[2016-12-16 13:53:05] VERBOSE[6631] res_pjsip_logger.c: <--- Received SIP
request (812 bytes) from UDP:11.111.11.11:5060 --->
UPDATE sip:222.222.222.22:5060 SIP/2.0
Via: SIP/2.0/UDP
11.111.11.11:5060;rport;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk
Max-Forwards: 70
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: sip:8xxx6yyy621 at txxx37.ru;tag=f30b688b-3358-4107-9992-fb6e3923bc15
Contact: "007" <sip:login at 11.111.11.11:5060;ob>
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
CSeq: 10073 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 90
Min-SE: 90
Content-Type: application/sdp
Content-Length:   271

v=0
o=- 3690874445 3690874446 IN IP4 11.111.11.11
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 96
c=IN IP4 11.111.11.11
b=TIAS:64000
a=rtcp:4023 IN IP4 11.111.11.11
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv

[2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP
response (910 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: <sip:8xxx6yyy621 at
txxx37.ru>;tag=f30b688b-3358-4107-9992-fb6e3923bc15
CSeq: 10073 UPDATE
Session-Expires: 90;refresher=uac
Require: timer
Contact: <sip:222.222.222.22:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Server: ruVoIP.net PBX
Content-Type: application/sdp
Content-Length:   242

v=0
o=- 3690874445 3690874448 IN IP4 222.222.222.22
s=ruVoIP.net PBX
c=IN IP4 222.222.222.22
t=0 0
m=audio 25094 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel
'PJSIP/login-0000027b' for lack of RTP activity in 10 seconds

[2016-12-16 13:53:15] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP
response (528 bytes) to UDP:11.111.11.11:5060 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: <sip:8xxx6yyy621 at
txxx37.ru>;tag=f30b688b-3358-4107-9992-fb6e3923bc15
CSeq: 10072 INVITE
Server: ruVoIP.net PBX
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
PRACK, MESSAGE, REFER, REGISTER
Reason: Q.850;cause=0
Content-Length:  0


[2016-12-16 13:53:16] VERBOSE[6631] res_pjsip_logger.c: <--- Received SIP
request (374 bytes) from UDP:11.111.11.11:5060 --->
ACK sip:8xxx6yyy621 at txxx37.ru SIP/2.0
Via: SIP/2.0/UDP
11.111.11.11:5060;rport;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql
Max-Forwards: 70
From: "007" <sip:login at
txxx37.ru>;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC
To: sip:8xxx6yyy621 at txxx37.ru;tag=f30b688b-3358-4107-9992-fb6e3923bc15
Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg
CSeq: 10072 ACK
Content-Length:  0

Jean Aunis

2016-Dec-19 08:33 UTC

head link

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

This means the remote end was not sending any audio stream, or the audio 
stream was not received by Asterisk. The problem may have many different 
reasons, but often it is a network-related issue.


Le 16/12/2016 ? 21:19, Dmitriy Serov a ?crit :> Today I faced a problem. Please help to solve this problem.
>
> Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware 
> v2.06(AAGJ.9)C1
>
> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk).
> Call using early media (183 Session in progress) and rtp_timeout=10.
> After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] 
> res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for
> lack of RTP activity in 10 seconds
>
> SIP dump is attached.
>
> According to [1] before called user agent send OK or ACK there is one 
> way SDP.
> In sip dump (attached) i didn't find such SIP packets. Whether that 
> call was canceled due to RTP inactivity?
>
> Any help is welcome.
>
> [1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt
>
>
>
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Dmitriy Serov

2016-Dec-19 10:36 UTC

head link

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

Yes, this means the remote end was not sending any audio stream.
But it shouldn't.
According to [1] before remote end send OK or ACK there is one way SDP, 
no any audio stream.
PJSIP channel (option rtp_timeout) does not take this one.

Isn't it a mistake? What could be workarounds?

19.12.2016 11:33, Jean Aunis ?????:>
> This means the remote end was not sending any audio stream, or the 
> audio stream was not received by Asterisk. The problem may have many 
> different reasons, but often it is a network-related issue.
>
>
> Le 16/12/2016 ? 21:19, Dmitriy Serov a ?crit :
>> Today I faced a problem. Please help to solve this problem.
>>
>> Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware 
>> v2.06(AAGJ.9)C1
>>
>> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip 
>> trunk).
>> Call using early media (183 Session in progress) and rtp_timeout=10.
>> After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] 
>> res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b'
for
>> lack of RTP activity in 10 seconds
>>
>> SIP dump is attached.
>>
>> According to [1] before called user agent send OK or ACK there is one 
>> way SDP.
>> In sip dump (attached) i didn't find such SIP packets. Whether that
>> call was canceled due to RTP inactivity?
>>
>> Any help is welcome.
>>
>> [1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt
>>
>>
>>
>
>
>
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asterisk users - Dec 2016 - 183 Session in Progress. Disconnecting channel for lack of RTP activity

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

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