Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
Joshua, issue has been filed. Thank you! https://issues.asterisk.org/jira/browse/ASTERISK-26689 03.01.2017 20:58, Joshua Colp пишет: On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: 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? Looking at the code we don't take that scenario into account it seems. Please file an issue[1] and we'll see if we can do something about it. [1] https://issues.asterisk.org/jira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: > 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? Looking at the code we don't take that scenario into account it seems. Please file an issue[1] and we'll see if we can do something about it. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 пишет: 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 à 21:19, Dmitriy Serov a écrit : 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-027b' 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 à 21:19, Dmitriy Serov a écrit : 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-027b' 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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-027b' 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 INVITE sip:8xxx6yyy...@txxx37.ru SIP/2.0 Via: SIP/2.0/UDP 11.111.11.11:5060;rport;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql Max-Forwards: 70 From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: sip:8xxx6yyy...@txxx37.ru Contact: "007" 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:8xxx6yyy...@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" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: 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" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15 CSeq: 10072 INVITE Server: ruVoIP.net PBX Contact: 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" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: sip:8xxx6yyy...@txxx37.ru;tag=f30b688b-3358-4107-9992-fb6e3923bc15 Contact: "007" 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" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15 CSeq: 10073 UPDATE Session-Expires: 90;refresher=uac Require: timer Contact: 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
