| Summary: | ASTERISK-09109: horrible audio with chan_cellphone | ||
| Reporter: | Roel Cuppen (cuppie) | Labels: | |
| Date Opened: | 2007-03-26 17:02:35 | Date Closed: | 2011-06-07 14:07:27 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Hi, Thanks for the great job ! I'm using the following -latest version of ubuntu feisty. (kernel 6.2.20) -AVM fritzfon 7150 as a sip device (G711) -Sitecom CN521 (CSR Chipset) (also an Etech dongle, and Eminent dongle) -Nokia 6230i -chan_cellphone-17.patch -current svn trunk of asterisk When calling from the sip device (avm 7150) to a destination via the cellphone (nokia 6230i) i'm getting lots of 'Overrun on sco_out_buf detected' like reporter stan_mahe has. In this case, audio from the sip phone to the destination (another cellphone) is horrible. Audio from the destination to the sip phone is perfect. Tested with 4 different types of BT dongles. Always the same. -Patching kernel with latest BT patches doesn't make sense -Recent firmware in nokia 6230i even doesn't make sense -Changing SCOMTU sizes doesn't make sense When changing the following in the source char sco_out_buf[352]; to char sco_out_buf[480]; The messages 'Overrun on sco_out_buf detected' are gone. But in this case , audio from the sip device to the destination is perfect. Audio from the destination to the sip device is not really bad , but could be a lot better. There are a lot of hickups, like rtp packets are dropped. Changing char sco_out_buf to a value lower than 480 , the overrun messages are back, and i got the primary problem. | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-03-26 17:13:34 Please stick to the original bug. | ||