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Hi
I use the GSM and G729 codec for an rtp Stack but know i have an Problem. I initialize the Codec at the beginning of the program and then i use for every rtp session these codecobject. When one rtp session is active it works fine but when i start an second rtp session i have a noise in the audiostream. The both rtp session are simultaniur active and the encoding is done one after one. Must I initilaize the codec every time when i encode an new rtp session?
Harald
I use the GSM and G729 codec for an rtp Stack but know i have an Problem. I initialize the Codec at the beginning of the program and then i use for every rtp session these codecobject. When one rtp session is active it works fine but when i start an second rtp session i have a noise in the audiostream. The both rtp session are simultaniur active and the encoding is done one after one. Must I initilaize the codec every time when i encode an new rtp session?
Harald
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Quoting - harald85
Hi
I use the GSM and G729 codec for an rtp Stack but know i have an Problem. I initialize the Codec at the beginning of the program and then i use for every rtp session these codecobject. When one rtp session is active it works fine but when i start an second rtp session i have a noise in the audiostream. The both rtp session are simultaniur active and the encoding is done one after one. Must I initilaize the codec every time when i encode an new rtp session?
Harald
I use the GSM and G729 codec for an rtp Stack but know i have an Problem. I initialize the Codec at the beginning of the program and then i use for every rtp session these codecobject. When one rtp session is active it works fine but when i start an second rtp session i have a noise in the audiostream. The both rtp session are simultaniur active and the encoding is done one after one. Must I initilaize the codec every time when i encode an new rtp session?
Harald
EachRTP session must be processed by separate USC codec instance.
Regards,
Vyacheslav
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