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

Fri Dec 16 14:19:48 CST 2016

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