aram
2005-Mar-19 20:30 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA; however X-Pro is having no problem. Similarly if the user is not authenticated - the call goes through fine. Is * timing out waiting for some response - if so how can we increase the timeout? We are also thinking of possibility that some port is closed over the long latency network that causing this problem - is that possible -if yes, which port would that be? Or is there another issue that we are not aware of? Thanks, Aram Debug (some parts of it) ====================== Sip read: INVITE sip:2002@xxx.xxx.xxx.xxx SIP/2.0 Content-Length: 360 Contact: <sip:2000@82.198.1.15:5060> Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15 Content-Type: application/sdp From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 CSeq: 1 INVITE Max-Forwards: 70 To: <sip:2002@xxx.xxx.xxx.xxx> Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f00007 9bb00000045 User-Agent: SJLabs-SJphone/1.40.258 ==== 12 headers, 16 lines Ignoring this request Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f000000244237ee60000052da000 00047 From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 To: <sip:2002@xxx.xxx.xxx.xxx>;tag=as0338b9e1 Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2002@xxx.xxx.xxx.xxx> Content-Length: 0 ======== toyyy.yyy.yyy.yyy:5060 In the log Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]: Stopping retransmission on '5CE9576F-92DB-4C4D-928E-DD704559AB38@82.198 Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 DEBUG[2325]: Ignoring too old packet packet 1 (expecting >= 2) Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 8414 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050319/261908ec/winmail.bin
aram
2005-Mar-22 01:43 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA; however X-Pro is having no problem. Similarly if the user is not authenticated - the call goes through fine. Is * timing out waiting for some response - if so how can we increase the timeout? We are also thinking of possibility that some port is closed over the long latency network that causing this problem - is that possible -if yes, which port would that be? Or is there another issue that we are not aware of? Thanks, Aram Debug (some parts of it) ====================== Sip read: INVITE sip:2002@xxx.xxx.xxx.xxx SIP/2.0 Content-Length: 360 Contact: <sip:2000@82.198.1.15:5060> Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15 Content-Type: application/sdp From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 CSeq: 1 INVITE Max-Forwards: 70 To: <sip:2002@xxx.xxx.xxx.xxx> Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f00007 9bb00000045 User-Agent: SJLabs-SJphone/1.40.258 ==== 12 headers, 16 lines Ignoring this request Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f000000244237ee60000052da000 00047 From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 To: <sip:2002@xxx.xxx.xxx.xxx>;tag=as0338b9e1 Call-ID: 959E39B1-B5BA-4F76-952B-192A9E4829EF@82.198.1.15 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2002@xxx.xxx.xxx.xxx> Content-Length: 0 ======== toyyy.yyy.yyy.yyy:5060 In the log Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]: Stopping retransmission on '5CE9576F-92DB-4C4D-928E-DD704559AB38@82.198 Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 DEBUG[2325]: Ignoring too old packet packet 1 (expecting >= 2) Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 8422 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050322/e6cd9630/winmail.bin
Craig
2005-Mar-22 02:07 UTC
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hi Aram, Think you will find it is something to do with ports etc.... We log soft phones, spa-841, grandstream and sipura ata through a satellite link that regularly drifts out to a latency of 1000ms ++ and do not see any problems with *. However some of the links we have used are into port blocking and stateful packet blocking. Suggest this would be most likely your cause. If you can, run it through a vpn over the same link and see if the problem persists, this will make life very simple. For those that think a 100ms is long latency, I have seen our link drift out to 1200ms latency with 100ms of jitter and still work, does start to get a bit crappy though! Craig. Message: 28 Date: Tue, 22 Mar 2005 00:43:35 -0800 From: "aram" <aram@hi-teck.com> Subject: [Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required To: <asterisk-users@lists.digium.com>, <asterisk-dev@lists.digium.com> Message-ID: <auto-000003091938@hi-teck.com> Content-Type: text/plain; charset="us-ascii" Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA; however X-Pro is having no problem. Similarly if the user is not authenticated - the call goes through fine. Is * timing out waiting for some response - if so how can we increase the timeout? We are also thinking of possibility that some port is closed over the long latency network that causing this problem - is that possible -if yes, which port would that be? Or is there another issue that we are not aware of? Thanks, Aram