[Freeswitch-dev] Help ! How to adjust the speex Codec 's parameter for my sip phone
hi, all I used my soft phone with speex phone in speex, 8000hz , my softphone ' codec is base on opal lib. but when I register it in freeswitch and call in a conference . I fine the voice quality is not good enough。 I did print the parameter in switch_speex_decode() function , I find that my softphone sent encoded_data_len=21. if I send frame as encoded_data_len=20, then the voice quality become nice. So ,who can tell me how to change the freeswitch's speex parameter, then I can operate the encoded_data_len=21 ? best regards . your freeswitch' fan. _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/soc...
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