global_s Posted October 12, 2012 Report Share Posted October 12, 2012 Hi, today a user notifed us that his call hanged after 35 seconds. here's the bye messagge; below you will find the complete log from the accident. It is not the first time that this happens, and everytime the bye message looks like this one BYE sip:50220@10.252.0.8:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-8xosfu11oled;rport From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:50220@10.246.0.106:2062;transport=tls;line=spr0c3mz>;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=2424,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=12478592 RTP-TxStat: Total_Tx_Pkts=2412,Tx_Pkts=2412,Remote_Tx_Pkts=1087924764 Proxy-Require: buttons Content-Length: 0 INVITE sip:393469482137@sipmsk.netkf.com;phone=yes SIP/2.0 Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-hbptx3xetzs2;rport From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes> Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:50220@10.246.0.106:2062;transport=tls;line=spr0c3mz>;reg-id=1 X-Serialnumber: 00041337BDA0 P-Key-Flags: keys="3" User-Agent: snom300/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 524 v=0 o=root 1039649131 1039649131 IN IP4 10.246.0.106 s=call c=IN IP4 10.246.0.106 t=0 0 m=audio 65418 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0eVzrnN67RykB8OhWPwo5RojUSKOqm1tmzC6bSOL a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [8] 2012/10/12 14:25:45: Packet authenticated by transport layer [8] 2012/10/12 14:25:45: Allocating for call port 48, SIP call id 3c2fac77d534-szxzmtu7s317 [9] 2012/10/12 14:25:45: UDP(IPv4): Opening socket on 0.0.0.0:10372 [9] 2012/10/12 14:25:45: UDP(IPv4): Opening socket on 0.0.0.0:10373 [9] 2012/10/12 14:25:45: UDP(IPv6): Opening socket on [::]:10372 [9] 2012/10/12 14:25:45: UDP(IPv6): Opening socket on [::]:10373 [8] 2012/10/12 14:25:45: Could not find a trunk (12 trunks) [9] 2012/10/12 14:25:45: Using outbound proxy sip:10.246.0.106:2062;transport=tls because of flow-label [9] 2012/10/12 14:25:45: Last message repeated 3 times [5] 2012/10/12 14:25:45: SIP Tx tls:10.246.0.106:2062: SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-hbptx3xetzs2;rport=2062 From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 1 INVITE Content-Length: 0 [7] 2012/10/12 14:25:45: Set packet length to 20 [6] 2012/10/12 14:25:45: Call-leg 48: Sending RTP for 3c2fac77d534-szxzmtu7s317 to 10.246.0.106:65418, codec not set yet [8] 2012/10/12 14:25:45: Incoming call: Request URI sip:393469482137@sipmsk.netkf.com;phone=yes, To is <sip:393469482137@sipmsk.netkf.com;phone=yes> [8] 2012/10/12 14:25:45: To is <sip:393469482137@sipmsk.netkf.com;phone=yes>, user 0, domain 2 [8] 2012/10/12 14:25:45: Set the To domain based on From user 50220@sipmsk.netkf.com [7] 2012/10/12 14:25:45: Call port 48: set_codecs for 3c2fac77d534-szxzmtu7s317 codecs "", codec_preference count 4 [9] 2012/10/12 14:25:45: Generating hf header using {trunk} [9] 2012/10/12 14:25:45: Generating ht header using {request-uri} [8] 2012/10/12 14:25:45: Play audio_moh/noise.wav, caching true [8] 2012/10/12 14:25:45: Allocating for call port 49, SIP call id c51fcce4@pbx [9] 2012/10/12 14:25:45: UDP(IPv4): Opening socket on 0.0.0.0:10934 [9] 2012/10/12 14:25:45: UDP(IPv4): Opening socket on 0.0.0.0:10935 [9] 2012/10/12 14:25:45: UDP(IPv6): Opening socket on [::]:10934 [9] 2012/10/12 14:25:45: UDP(IPv6): Opening socket on [::]:10935 [7] 2012/10/12 14:25:45: Call port 49: set_codecs for c51fcce4@pbx codecs "0", codec_preference count 2 [8] 2012/10/12 14:25:45: call port 49: state code from 0 to 100 [9] 2012/10/12 14:25:45: Call port 49: update_codecs for c51fcce4@pbx: adding codec pcmu/8000 to available list [9] 2012/10/12 14:25:45: Call port 49: update_codecs for c51fcce4@pbx: codec_preference size 2, available codecs size 2 [9] 2012/10/12 14:25:45: Resolve 752110: url sip:sip.outobouond.com:5060 [9] 2012/10/12 14:25:45: Resolve 752110: a udp sip.outobouond.com 5060 [8] 2012/10/12 14:25:45: DNS: Request sip.outobouond.com from server 8.8.8.8 [8] 2012/10/12 14:25:46: call port 48: state code from 0 to 183 [7] 2012/10/12 14:25:46: Set packet length to 20 [9] 2012/10/12 14:25:46: Call port 48: update_codecs for 3c2fac77d534-szxzmtu7s317: adding codec pcmu/8000 to available list [9] 2012/10/12 14:25:46: Call port 48: update_codecs for 3c2fac77d534-szxzmtu7s317: adding codec g722/8000 to available list [9] 2012/10/12 14:25:46: Call port 48: update_codecs for 3c2fac77d534-szxzmtu7s317: adding codec g729/8000 to available list [9] 2012/10/12 14:25:46: Call port 48: update_codecs for 3c2fac77d534-szxzmtu7s317: codec_preference size 4, available codecs size 4 [6] 2012/10/12 14:25:46: Call-leg 48: Codec pcmu/8000 is chosen for call id 3c2fac77d534-szxzmtu7s317 [5] 2012/10/12 14:25:46: set codec: codec pcmu/8000 is set to call-leg 48 [5] 2012/10/12 14:25:46: SIP Tx tls:10.246.0.106:2062: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-hbptx3xetzs2;rport=2062 From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 1 INVITE Contact: <sip:50220@10.252.0.8:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 400 v=0 o=- 946127172 946127172 IN IP4 10.252.0.8 s=- c=IN IP4 10.252.0.8 t=0 0 m=audio 10372 RTP/AVP 0 9 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:h5sDUERDsscbXwl21rppk9AHdQVQVP7XtpuVlnSF a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [8] 2012/10/12 14:25:46: DNS: Add A sip.outobouond.com 43.12.112.201 (ttl=269) [9] 2012/10/12 14:25:46: DNS: erasing A sip.outobouond.com, id 1594 retry count 0, [9] 2012/10/12 14:25:46: Resolve 752110: a udp sip.outobouond.com 5060 [9] 2012/10/12 14:25:46: Resolve 752110: udp 43.12.112.201 5060 [5] 2012/10/12 14:25:46: SIP Tx udp:43.12.112.201:5060: INVITE sip:00393469482137@sip.outobouond.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-c3248993c23a181805760b408158ffd7;rport From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com> Call-ID: c51fcce4@pbx CSeq: 4405 INVITE Max-Forwards: 70 Contact: <sip:99051000170744@1.172.102.70:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids P-Charging-Vector: icid-value=;icid-generated-at=1.172.102.70;orig-ioi=sipmsk.netkf.com Content-Type: application/sdp Content-Length: 237 v=0 o=- 615369533 615369533 IN IP4 1.172.102.70 s=- c=IN IP4 1.172.102.70 t=0 0 m=audio 10934 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/10/12 14:25:46: SIP Rx tls:10.246.0.106:2062: PRACK sip:50220@10.252.0.8:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-80lo3b6e7p9h;rport From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:50220@10.246.0.106:2062;transport=tls;line=spr0c3mz>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [8] 2012/10/12 14:25:46: Packet authenticated by transport layer [5] 2012/10/12 14:25:46: SIP Tx tls:10.246.0.106:2062: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-80lo3b6e7p9h;rport=2062 From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 2 PRACK Contact: <sip:50220@10.252.0.8:5061;transport=tls> User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [8] 2012/10/12 14:25:46: SRTP MAC mismatch: f2f14e72 != 4f4d0000 [7] 2012/10/12 14:25:46: Discard SRTCP packet from 10.246.0.106:65419 with wrong MAC [5] 2012/10/12 14:25:46: SIP Rx udp:43.12.112.201:5060: SIP/2.0 100 Trying From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com> Call-ID: c51fcce4@pbx CSeq: 4405 INVITE Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-c3248993c23a181805760b408158ffd7;rport=5060 Contact: <sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp> Content-Length: 0 [9] 2012/10/12 14:25:46: Message repetition, packet dropped [5] 2012/10/12 14:25:46: SIP Rx udp:43.12.112.201:5060: SIP/2.0 407 Proxy Authentication Required From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4405 INVITE Proxy-Authenticate: Digest realm="sip.outobouond.com", nonce="50780d3700000cc1e717955992bb1bdb74afa0bcdce83793", algorithm=MD5 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-c3248993c23a181805760b408158ffd7;rport=5060 Content-Length: 0 [8] 2012/10/12 14:25:46: Answer challenge with username 99051000170744 [9] 2012/10/12 14:25:46: Resolve 752113: udp 43.12.112.201 5060 udp:1 [9] 2012/10/12 14:25:46: Resolve 752114: udp 43.12.112.201 5060 udp:1 [5] 2012/10/12 14:25:46: SIP Tx udp:43.12.112.201:5060: INVITE sip:00393469482137@sip.outobouond.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-2c319aeedb431ad029bd3103168d7d4f;rport From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com> Call-ID: c51fcce4@pbx CSeq: 4406 INVITE Max-Forwards: 70 Contact: <sip:99051000170744@1.172.102.70:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids P-Charging-Vector: icid-value=;icid-generated-at=1.172.102.70;orig-ioi=sipmsk.netkf.com Proxy-Authorization: Digest realm="sip.outobouond.com",nonce="50780d3700000cc1e717955992bb1bdb74afa0bcdce83793",response="4e7cb36f06601bfda907d0ae5a634e33",username="99051000170744",uri="sip:00393469482137@sip.outobouond.com;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 237 v=0 o=- 615369533 615369533 IN IP4 1.172.102.70 s=- c=IN IP4 1.172.102.70 t=0 0 m=audio 10934 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [9] 2012/10/12 14:25:46: Message repetition, packet dropped [5] 2012/10/12 14:25:46: SIP Rx udp:43.12.112.201:5060: SIP/2.0 100 Trying From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4406 INVITE Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-2c319aeedb431ad029bd3103168d7d4f;rport=5060 Contact: <sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp> Content-Length: 0 [9] 2012/10/12 14:25:46: Message repetition, packet dropped [8] 2012/10/12 14:25:48: Packet authenticated by transport layer [8] 2012/10/12 14:25:49: Last message repeated 2 times [9] 2012/10/12 14:25:49: Resolve 752117: udp 23.23.53.51 2048 [8] 2012/10/12 14:25:49: Packet authenticated by transport layer [8] 2012/10/12 14:25:51: Trunk 8: Preparing for re-registration [8] 2012/10/12 14:25:51: Trunk Xeloq_redirect: Sending registration to sip.goandcall.com:6060 [9] 2012/10/12 14:25:51: Resolve 752119: udp 81.26.212.150 6060 [9] 2012/10/12 14:25:51: Message repetition, packet dropped [8] 2012/10/12 14:25:51: Answer challenge with username 737629 [9] 2012/10/12 14:25:51: Resolve 752120: udp 81.26.212.150 6060 udp:1 [9] 2012/10/12 14:25:51: Message repetition, packet dropped [9] 2012/10/12 14:25:51: Last message repeated 2 times [8] 2012/10/12 14:25:51: Trunk 8: setup callback to send re-registration after 1800 seconds [8] 2012/10/12 14:25:54: Packet authenticated by transport layer [9] 2012/10/12 14:25:54: Resolve 752122: udp 84.199.103.82 5060 [5] 2012/10/12 14:25:55: SIP Rx udp:43.12.112.201:5060: SIP/2.0 180 Ringing From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4406 INVITE User-Agent: SipGW 9 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-2c319aeedb431ad029bd3103168d7d4f;rport=5060 Contact: <sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp> Content-Length: 0 [8] 2012/10/12 14:25:55: Play audio_it/ringback.wav, caching true [8] 2012/10/12 14:25:55: call port 48: state code from 183 to 183 [5] 2012/10/12 14:25:55: SIP Rx udp:43.12.112.201:5060: SIP/2.0 180 Ringing From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4406 INVITE User-Agent: SipGW 9 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-2c319aeedb431ad029bd3103168d7d4f;rport=5060 Contact: <sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp> Content-Length: 0 [8] 2012/10/12 14:25:55: Packet authenticated by transport layer [8] 2012/10/12 14:25:59: Last message repeated 3 times [6] 2012/10/12 14:25:59: Call-leg 49: Sending RTP for c51fcce4@pbx to 43.12.112.201:28014, codec not set yet [5] 2012/10/12 14:25:59: SIP Rx udp:43.12.112.201:5060: SIP/2.0 200 OK From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4406 INVITE Allow: INVITE,ACK,CANCEL,OPTIONS,BYE User-Agent: SipGW 9 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-2c319aeedb431ad029bd3103168d7d4f;rport=5060 Contact: <sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp> Content-Type: application/sdp Content-Length: 196 v=0 o=- 615369533 615369533 IN IP4 43.12.112.201 s=outobouond call c=IN IP4 43.12.112.201 t=0 0 m=audio 28014 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 pcmu/8000 a=ptime:20 [7] 2012/10/12 14:25:59: Call c51fcce4@pbx: Clear last INVITE [7] 2012/10/12 14:25:59: Set packet length to 20 [6] 2012/10/12 14:25:59: Call-leg 49: Codec pcmu/8000 is chosen for call id c51fcce4@pbx [5] 2012/10/12 14:25:59: set codec: codec pcmu/8000 is set to call-leg 49 [9] 2012/10/12 14:25:59: Resolve 752126: aaaa udp 43.12.112.201 5060 [9] 2012/10/12 14:25:59: Resolve 752126: a udp 43.12.112.201 5060 [9] 2012/10/12 14:25:59: Resolve 752126: udp 43.12.112.201 5060 [5] 2012/10/12 14:25:59: SIP Tx udp:43.12.112.201:5060: ACK sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-54a012fa91696933dcaa209f14aebc66;rport From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4406 ACK Max-Forwards: 70 Contact: <sip:99051000170744@1.172.102.70:5060;transport=udp> P-Charging-Vector: icid-value=;icid-generated-at=1.172.102.70;orig-ioi=sipmsk.netkf.com Content-Length: 0 [7] 2012/10/12 14:25:59: Determine pass-through mode after receiving response [8] 2012/10/12 14:25:59: call port 49: state code from 100 to 200 [8] 2012/10/12 14:25:59: call port 48: state code from 183 to 200 [5] 2012/10/12 14:25:59: SIP Tx tls:10.246.0.106:2062: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-hbptx3xetzs2;rport=2062 From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 1 INVITE Contact: <sip:50220@10.252.0.8:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 400 v=0 o=- 946127172 946127172 IN IP4 10.252.0.8 s=- c=IN IP4 10.252.0.8 t=0 0 m=audio 10372 RTP/AVP 0 9 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:h5sDUERDsscbXwl21rppk9AHdQVQVP7XtpuVlnSF a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/10/12 14:25:59: 3c2fac77d534-szxzmtu7s317: RTP pass-through mode [7] 2012/10/12 14:25:59: c51fcce4@pbx: RTP pass-through mode [5] 2012/10/12 14:25:59: SIP Rx tls:10.246.0.106:2062: ACK sip:50220@10.252.0.8:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-rpwhy3qfzgtw;rport From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:50220@10.246.0.106:2062;transport=tls;line=spr0c3mz>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [8] 2012/10/12 14:25:59: Packet authenticated by transport layer [8] 2012/10/12 14:26:06: Last message repeated 3 times [9] 2012/10/12 14:26:06: Resolve 752130: udp 23.23.53.51 2048 [8] 2012/10/12 14:26:07: Packet authenticated by transport layer [8] 2012/10/12 14:26:11: Last message repeated 5 times [8] 2012/10/12 14:26:11: Trunk 22: Preparing for re-registration [8] 2012/10/12 14:26:11: Trunk outobouond: Sending registration to sip.outobouond.com:5060 [9] 2012/10/12 14:26:11: Resolve 752136: udp 43.12.112.201 5060 [8] 2012/10/12 14:26:11: Trunk 22: setup callback to send re-registration after 150 seconds [8] 2012/10/12 14:26:13: Packet authenticated by transport layer [9] 2012/10/12 14:26:16: Resolve 752138: udp 23.23.53.51 2048 [9] 2012/10/12 14:26:24: Resolve 752139: udp 84.199.103.82 5060 [8] 2012/10/12 14:26:30: Packet authenticated by transport layer [9] 2012/10/12 14:26:31: Resolve 752141: udp 23.23.53.51 2048 [5] 2012/10/12 14:26:34: SIP Rx tls:10.246.0.106:2062: BYE sip:50220@10.252.0.8:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-8xosfu11oled;rport From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:50220@10.246.0.106:2062;transport=tls;line=spr0c3mz>;reg-id=1 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=2424,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=12478592 RTP-TxStat: Total_Tx_Pkts=2412,Tx_Pkts=2412,Remote_Tx_Pkts=1087924764 Proxy-Require: buttons Content-Length: 0 [8] 2012/10/12 14:26:34: Packet authenticated by transport layer [5] 2012/10/12 14:26:34: SIP Tx tls:10.246.0.106:2062: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.246.0.106:2062;branch=z9hG4bK-8xosfu11oled;rport=2062 From: "US Group" <sip:50220@sipmsk.netkf.com>;tag=f6musjqxr5 To: <sip:393469482137@sipmsk.netkf.com;phone=yes>;tag=cfa5aeec1f Call-ID: 3c2fac77d534-szxzmtu7s317 CSeq: 3 BYE Contact: <sip:50220@10.252.0.8:5061;transport=tls> User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [7] 2012/10/12 14:26:34: c51fcce4@pbx: Media-aware pass-through mode [8] 2012/10/12 14:26:34: Clearing call port 48, SIP call id 3c2fac77d534-szxzmtu7s317 [8] 2012/10/12 14:26:34: call port 49: state code from 200 to 486 [9] 2012/10/12 14:26:34: Resolve 752143: aaaa udp 43.12.112.201 5060 [9] 2012/10/12 14:26:34: Resolve 752143: a udp 43.12.112.201 5060 [9] 2012/10/12 14:26:34: Resolve 752143: udp 43.12.112.201 5060 [5] 2012/10/12 14:26:34: SIP Tx udp:43.12.112.201:5060: BYE sip:00393469482137@sip.outobouond.com:5060;user=phone;maddr=43.12.112.201;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-d62c98b4a591dd9217462b9b7def302c;rport From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4407 BYE Max-Forwards: 70 Contact: <sip:99051000170744@1.172.102.70:5060;transport=udp> P-Charging-Vector: icid-value=;icid-generated-at=1.172.102.70;orig-ioi=sipmsk.netkf.com Content-Length: 0 [8] 2012/10/12 14:26:34: Remove leg 7031: call port 48, SIP call id 3c2fac77d534-szxzmtu7s317 [8] 2012/10/12 14:26:34: Hangup: Call 48 not found [8] 2012/10/12 14:26:34: Last message repeated 2 times [5] 2012/10/12 14:26:34: SIP Rx udp:43.12.112.201:5060: SIP/2.0 200 OK From: "99051000170744" <sip:99051000170744@sip.outobouond.com>;tag=123821746 To: <sip:00393469482137@sip.outobouond.com>;tag=c990d13f-13c4-50780d18-5769a3c1-6569fae3 Call-ID: c51fcce4@pbx CSeq: 4407 BYE Via: SIP/2.0/UDP 1.172.102.70:5060;branch=z9hG4bK-d62c98b4a591dd9217462b9b7def302c;rport=5060 Content-Length: 0 [7] 2012/10/12 14:26:34: Call c51fcce4@pbx: Clear last request [5] 2012/10/12 14:26:34: BYE Response: Terminate c51fcce4@pbx [8] 2012/10/12 14:26:34: Clearing call port 49, SIP call id c51fcce4@pbx [8] 2012/10/12 14:26:34: Remove leg 7032: call port 49, SIP call id c51fcce4@pbx [9] 2012/10/12 14:26:36: Resolve 752144: udp 23.23.53.51 2048 [8] 2012/10/12 14:26:36: Packet authenticated by transport layer [8] 2012/10/12 14:26:53: Last message repeated 5 times [9] 2012/10/12 14:26:53: Resolve 752150: udp 84.199.103.82 5060 [9] 2012/10/12 14:26:54: Resolve 752151: udp 23.23.53.51 2048 [8] 2012/10/12 14:26:55: Packet authenticated by transport layer [8] 2012/10/12 14:27:02: Last message repeated 4 times [9] 2012/10/12 14:27:02: Resolve 752156: udp 23.23.53.51 2048 [8] 2012/10/12 14:27:02: Packet authenticated by transport layer [8] 2012/10/12 14:27:06: Last message repeated 3 times [9] 2012/10/12 14:27:06: Resolve 752160: udp 23.23.53.51 2048 [8] 2012/10/12 14:27:15: Packet authenticated by transport layer [8] 2012/10/12 14:27:20: Last message repeated 3 times Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 12, 2012 Report Share Posted October 12, 2012 User-Agent: snom300/8.4.32 RTP-RxStat: Total_Rx_Pkts=2424,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=12478592 RTP-TxStat: Total_Tx_Pkts=2412,Tx_Pkts=2412,Remote_Tx_Pkts=1087924764 The SIP signalling looks "beautiful" as far as I can see. My worry is that the phone obviously did not receive any (valid) packet. Why the phone thinks that 1 billion packets have been sent from remote is another question (I suspect a minor glitch in the phone). Because the phone is in the same subnet like the PBX, I don't think we have the usual firewall/NAT problems. My feeling is that there is something wrong with the SRTP. Anything in the phone that hints after MAC check problems? If you can, try to use TCP transport layer for that phone; then the RTP should be unencrypted and we can rule out that SRTP is the problem. Quote Link to comment Share on other sites More sharing options...
global_s Posted October 28, 2012 Author Report Share Posted October 28, 2012 The SIP signalling looks "beautiful" as far as I can see. My worry is that the phone obviously did not receive any (valid) packet. Why the phone thinks that 1 billion packets have been sent from remote is another question (I suspect a minor glitch in the phone). Because the phone is in the same subnet like the PBX, I don't think we have the usual firewall/NAT problems. My feeling is that there is something wrong with the SRTP. Anything in the phone that hints after MAC check problems? If you can, try to use TCP transport layer for that phone; then the RTP should be unencrypted and we can rule out that SRTP is the problem. Hi, I disabled SRTP but the problem is still present. The PBX is not in the same subnet as the phone. Phone is in 10.246.0.0/24 while pbx is 10.252.0.0/24 they are connected through a MPLS network. I don't understand why it does not work only sometimes. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 29, 2012 Report Share Posted October 29, 2012 Hmm. Seems to be tricky. Are IP address conflicts possible? Or are certain ports blocked somewhere down the routing path? MPLS sounds very good in principle; maybe you can double check if the routing from the PBX to the phone works like a no-brainer. Quote Link to comment Share on other sites More sharing options...
global_s Posted December 4, 2012 Author Report Share Posted December 4, 2012 Hmm. Seems to be tricky. Are IP address conflicts possible? Or are certain ports blocked somewhere down the routing path? MPLS sounds very good in principle; maybe you can double check if the routing from the PBX to the phone works like a no-brainer. Update I switched Follow RTP off and for the moment we haven't had any issue. I did not say earlier that on the same subnet there is another SIP PBX. Maybe snomONE listened to the RTP sent from the other PBX and disconnected the RTP that he was supposed to stick with. Could this be the reason of the disconnection? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 4, 2012 Report Share Posted December 4, 2012 Yes that can be a reason. According to IETF it is okay if the RTP stream comes from a different location than where it is being sent to. Although it is very rate and extremly NAT unfriendly; but in theory it can happen. Quote Link to comment Share on other sites More sharing options...
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