如何最小化或禁用从freeswitch到星号的编解码器转码？
[英]how to minimize or disable codec transcoding from freeswitch to asterisk?
问题描述
i want to pass call from freeswitch to asterisk without transcoding, freeswitch(Version 1.4.6 32bit) and asterisk-11.7.0 are installed at same machine, in freeswitch call codec description is write_codec=L16 write_rate=16000 write_bit_rate=256000, in freeswitch i have tried 
我想将呼叫从freeswitch传递到星号而不进行转码，freeswitch（版本1.4.6 32bit）和asterisk-11.7.0安装在同一台机器上，在freeswitch呼叫编解码器中的描述是write_codec = L16 write_rate = 16000 write_bit_rate = 256000，在freeswitch中我努力了
<param name="disable-transcoding" value="true"/>
then asterisk displays 然后显示星号
Found RTP audio format 98
Found RTP audio format 13
Found audio description format L16 for ID 98
chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer!
I think freeswitch internally using L16 and asterisk is using SLIN , 我认为内部使用L16和星号的freeswitch使用SLIN ，
How to minimize or disable codec transcoding in both freeswitch and asterisk? 如何最小化或禁用freeswitch和星号中的编解码器转码？
1 个解决方案
解决方案1
0 已采纳 2015-09-20 21:02:39
You also can use g722 or any other narow band uncompressed codecs. 您还可以使用g722或任何其他narow band未压缩的编解码器。
All uncompressed codecs requires very low cpu to do transcode into any other uncompressed(much less then handling of stream) 所有未压缩的编解码器都需要非常低的cpu才能将代码转码为任何其他未压缩的代码（比处理流要少得多）
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