call-id doesn't match? SIP/2.0 200 OK ... Call-ID: [email protected] ...
ACK sip:[email protected] SIP/2.0 ... Call-ID: [email protected] ... I'm not sure, but I think that the part after the '@' must also match throughout the dialog. A Grandstream bug? -----Original Message----- From: [email protected] on behalf of Alejandro Recarey Sent: Fri 4/23/2010 6:36 PM To: Asterisk Users Mailing List Subject: [asterisk-users] Asterisk not recognizing ACK from an OK message?Help debuging SIP retransmit problem Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes Asterisk retransmit the OK message 6 times, after which it drops the call (after exactly 20 seconds). The strange thing is that the "branch" parameter of the ACK is different than the branch parameter from the OK it is replying to, however, this seems to be normal behaviour as specified in the RFC for an ACK that is sent in response to a 200 message. The full SIP dialog is at http://pastie.org/private/nybdytnfyfenovpwfywcya so as to not clutter the email, but I have included the highlights below: >>> The call was ringing and is now answered: <--- Reliably Transmitting (NAT) to 82.158.83.xxx:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 INVITE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/800902-00001794 and SIP/130.117.110.21-00001795 >>> ATA ACK's the OK message: <--- SIP read from UDP://82.158.83.xxx:5062 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1788524356;rport From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 ACK Contact: <sip:[email protected]:5062;user=phone> Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1C 1.0.1.57 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 >>> Asterisk does not recognize and retransmits <-------------> --- (12 headers 0 lines) --- Retransmitting #1 (NAT) to 82.158.83.xxx:5062: SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 INVITE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv >>> ACK is received again <--- SIP read from UDP://82.158.83.xxx:5062 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1788524356;rport From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 ACK Contact: <sip:[email protected]:5062;user=phone> Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1C 1.0.1.57 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 ............... (cut) >>> The retransmits happen 6 times and then: <-------------> --- (12 headers 0 lines) --- Retransmitting #6 (NAT) to 82.158.83.xxx:5062: SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 INVITE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP://82.158.83.xxx:5062 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1788524356;rport From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 31 ACK Contact: <sip:[email protected]:5062;user=phone> Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1C 1.0.1.57 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP://82.158.83.xxx:5062 ---> <-------------> [Apr 23 02:37:15] WARNING[3202]: chan_sip.c:3396 retrans_pkt: Maximum retries exceeded on transmission [email protected] for seqno 31 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 23 02:37:15] WARNING[3202]: chan_sip.c:3423 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (default, 6615xxxxx, 27) exited non-zero on 'SIP/800902-00001794' Really destroying SIP dialog '[email protected]' Method: INVITE <--- SIP read from UDP://82.158.83.xxx:5062 ---> BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1883566966;rport From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 32 BYE Contact: <sip:[email protected]:5062;user=phone> Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1C 1.0.1.57 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- Transmitting (no NAT) to 82.158.83.xxx:5062 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1883566966;received=82.158.83.xxx;rport=5062 From: "800902" <sip:[email protected];user=phone>;tag=467506068 To: <sip:[email protected];user=phone>;tag=as2e12c791 Call-ID: [email protected] CSeq: 32 BYE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer ontent-Length: 0 <------------> Thank you for your help! Alex -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
<<winmail.dat>>
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
