I'm trying to convert a proprietary audio stream into sip/rtp compatible. At this point, i'm able to pass sip negotiation with FS, and trying to stream PCMU/8000 codec. But something not right on my end, and FS complained about the ptime settings.<div>
I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right amount? appreciate if anyone could point some info on these topics.</div><div><br></div><div>Thanks,</div><div>JK<br><div><br></div><div>===============</div>
<div><div><b>2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use ptime 20 but what they meant to say was 40</b></div><div><b>This issue has so far been identified to happen on the following broken platforms/devices:</b></div>
<div><b>Linksys/Sipura aka Cisco</b></div><div><b>ShoreTel</b></div><div><b>Sonus/L3</b></div><div><b>We will try to fix it but some of the devices on this list are so broken,</b></div><div><b>who knows what will happen..</b></div>
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