[asterisk-bugs] [Asterisk 0010229]: Sound problems when bridging sip channels with packetization level different than 20ms

[noreply at bugs.digium.com]

The following issue has been CLOSED 
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http://bugs.digium.com/view.php?id=10229 
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Reported By:                gspiliot
Assigned To:                dbowerman
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Project:                    Asterisk
Issue ID:                   10229
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 416 
Disclaimer on File?:        No 
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             07-18-2007 11:02 CDT
Last Modified:              09-28-2007 19:40 CDT
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Summary:                    Sound problems when bridging sip channels with
packetization level different than 20ms
Description: 
When bridging chan_mobile with another sip channel which, for example,
receives packets with 90ms size instead of the default 20ms then the sound
coming out chan_mobile is choppy. It seems that repacketization or the rtp
stream is needed if chan_mobile works with fixed 20ms packets.
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Relationships       ID      Summary
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related to          0010076 One-way audio (one-way perfect and one-...
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---------------------------------------------------------------------- 
 dbowerman - 09-28-07 19:40  
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no feedback, cant reproduce with latest trunk. 

Issue History 
Date Modified   Username       Field                    Change               
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09-28-07 19:40  dbowerman      Note Added: 0071192                          
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