![]() if it wasn't, then send a dummy packet with the same timestamp as the last one to mark the end of the last frame? sounds a bit hacky, but not sure how else to make the jitter buffer happy here. i think there are a couple options:ġ) send the resumed packets with the same timestamp as the old one.this gets weird though, as with h264 it could throw things off with regards to the market bit, since the jitter buffer will be looking for the next packet with one to know when the frame ends.Ģ) send the resumed packets with the next anticipated timestamp: this won't work as it'll cause the same slow switching problem we observed when filing the bug.ģ) remember if the last packet sent was the end of a frame or not.if it was, can send the new data with the next timestamp without tweaking anything. In order to address the issue above (for h264, at least) we'd need to make the jitter buffer happy so it flushes correctly. stream 1 starts being forwarded again, sequence and timestamp are obviously much further ahead but are re-written to match Plus, add Jitsi meetings to your calendar and start them with one click. Connect your calendar to view all your meetings in Jitsi Meet. now a period where stream 1 isn't being forwarded. Or book a meeting URL in advance where you are the only moderator. Stream 1 stops being forwarded, last packets were: are you doing anything to handle this scenario? Reply to this email directly or view it on looks like this may fix for us, which would be great. You are receiving this because you are subscribed to this thread. There is no check that what getLength returns is: GetVersion checks whether the length is at least 1 byte getLength checks whether the length is at least 4 bytes. + int pktLen = RTCPHeaderUtils.getLength(buf, off, len) + int version = RTCPHeaderUtils.getVersion(buf, off, len) + public static boolean isValid(byte buf, int off, int len) ![]() Check that the microphone meter shows movement when you make noise. + * true if the RTCP packet is valid, false otherwise. Troubleshooting No sound If you are currently on a voice call take a look at the Call window. + * len the number of bytes in buffer which constitute the actual This is working fine and the call goes through. My use case is so that first a user creates a conference through jitsi-meet, and then dials an external participant over Jigasi. Desirable Births Register Sure Jitsi Meet. Hi, I am running jitsi-meet with jigasi and Asterisk, and I am running into a delay after hanging up. ![]() + * off the offset in the byte buffer where the RTCP header starts. Inmate Visitation Eligible: NO The inmate is temporarily ineligible for visitation. + * buf the byte buffer that contains the RTCP header. ![]()
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