Hi all,
I have studied this problem for awhile and feel very frustrated. Really need a hand 
Here is my situation:
My asterisk version is 1.6.2.24 and is installed on an embedded board(MT7620). The music is recorded with PCM format , 8K sample/sec, wav extension but it cannot be played smoothly when the SIP_Client is in hold.
Mostly people said you should enable internal_timing=yes in asterisk.conf. I have do so. But the situation is not getting better:
[directories]
astetcdir=>/etc/asterisk
astmoddir=>/apps/usr/lib/asterisk/modules
astvarlibdir=>/var/lib/asterisk
astagidir=>/var/lib/asterisk/agi-bin
astspooldir=>/var/spool/asterisk
astrundir=>/var/run
astlogdir=>/var/log/asterisk
[options]
internal_timing=yes
At first, I thought the problem is caused by no timing interface is not enable in asterisk because I dont have DAHDI. Thererfore, I use res_pthread_timing (I monitor pthread using core show threads, I can see it running) but the music is still choppy.
Then, I trace the code to res_musiconhold.c and measure the time in bridge_p2p_loop. I found the time interval between sending the music data is too long (about 30ms ~ 50ms). It is a little weird that asterisk does not generate music at 20ms pace when the music is 8K sample/sec.
Thanks for sparing your time reading my problem!