I am creating a translation module for AMR-WB. In one scenario on the SIP/SDP layer, a higher ptime was negotiated than the default one. For example, 60ms were negotiated instead of the AMR default 20ms. Now, Asterisk should send three frames per RTP packet. I try to play one of the recorded voices (slin8). Asterisk sends 320 samples to my translation module; the default for 20ms packetization. My translation module has to wait 960 samples to create the frames.
Which structure do I have to query: How do know the ptime, there in such a transcoding module? This was available in Asterisk 11 via ast_format->cur_ms. How do I access this information in Asterisk 13|master? I am asking asterisk-dev, because all transcoding modules might be affected. At least with AMR-WB, I have to code/build those frames within the module because RFC 4867 mandates a special header (section 4.3.5.2 and 4.4.5.1). At first glance, this could be an architectural issue. Therefore, I am asking for advice how to approach this: Simply, re-adding "cur_ms" to ast_format? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
