| Summary: | ASTERISK-01456: timestamps slip going sip to sip with latest cvs-head. | ||
| Reporter: | testmaster (testmaster) | Labels: | |
| Date Opened: | 2004-04-22 14:35:18 | Date Closed: | 2011-06-07 14:10:42 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) log.txt ( 1) output.txt | |
| Description: | I have a incoming Sip call from a did provider. And the sound Is really bad. I keep getting a fax/data Sound when ever someone places a inbound call to me.. I can`t for the life of me figure out a way to fix this... I included a log.txt file on the sip debug | ||
| Comments: | By: Brian West (bkw918) 2004-04-22 16:25:39 Go get ethereal and collect timestamp info please. tethereal -n not port ssh thats what I use By: Brian West (bkw918) 2004-04-22 18:55:19 I can confirm that the timestamps are slipping ..related to bug 1374. By: Brian West (bkw918) 2004-04-23 01:33:59 I have spent the better part of an hour trying to recreate this between two asterisk boxes with 729 and 711 ... its been FLAWLESS. By: Mark Spencer (markster) 2004-04-23 01:37:21 There's also no explanation of what IP is what in this attached file By: Mark Spencer (markster) 2004-04-24 11:23:17 Please supply some additional information if you're still having this problem, such as an explanation of what's what in the attached debug. I also notice there is an iax portion of this call. There was an iax trunking timestmap slip issue that has been resolved as of yesterday. Since we are so close to having our first 1.0 release candidate of Asterisk, we are asking poeple with "MAJOR", "BLOCK" or "CRASH" bugs to be as prompt as possible in responding to requests here, especially on issues that cannot be duplicated by the bug marshals. By: Brian West (bkw918) 2004-04-24 12:03:12 TestMaster find me on irc if this is still a problem. I updated both ends of my setup and tried to recreate this and coudln't. By: testmaster (testmaster) 2004-04-24 14:57:28 I`m sorry for reopening this bug.. But here is all the ip address.. Also the problem is still the same SIP phone 64.119.111.243 Asterisk -> 64.187.19.238 And the Sentito server ip ->64.187.19.237 This problem i`m having is not due to the iax call you seen. It happens from Did to sip. Only i do not have this problem when i`m going did to iax and the output.txt file I believe i called it is. the out put bkw asked for. And i`m running the current cvs and the problem is still there.. If you need access to that server at all you are welcome to email me. or talk to me on irc. Sorry thats all the information i have right now. By: Brian West (bkw918) 2004-04-24 15:55:40 I have this feeling the non-asterisk side is using G729 30ms because I see 240ms jumps that do support that theory. Asterisk only does G729 20ms. bkw By: Brian West (bkw918) 2004-04-24 16:18:20 Downgrade to minor pending more info. By: Brian West (bkw918) 2004-04-29 00:21:23 VAD and 30ms vs 20ms codec drama. Asterisk only supports 20ms g729 | ||