[b]1.can't make call again and again.[/b] (only one successful in five call)
why: did not update inviteDialog after a call in time. and then when oncalling again,it would can't handle a strange inviteDialog.
fixed: update inviteDialog after a call in time, and try to wait before a new Dialog.
[b]2. RED5Phone to RED5Phone the call quality is very bad[/b]
why: there are some codec problem between rtmpuser and rtmpuser(send and receive). I will fix it later( for handle flash to sip phone)
fixed(new direction): before stream callflow : web publish --RTMP-> red5 server --RTMP->red5phone server --RTP-> red5phone server --RTMP-> red5 server --RTMP -> web receiver
fixed stream callflow: web publish --RTMP-> red5 server--RTMP -> web receiver
[b]3. ONEWAY AUDIO in Transfer [/b]
why: can't exchange RTPStream 's remoteAddr correctly.
fixed: exchange RTPStream 's remoteAddr correctly. in the new direction(delete RTMP --> RTP --> RTP -->RTMP) just exchange their publish ID correctly.
why: did not update inviteDialog after a call in time. and then when oncalling again,it would can't handle a strange inviteDialog.
fixed: update inviteDialog after a call in time, and try to wait before a new Dialog.
[b]2. RED5Phone to RED5Phone the call quality is very bad[/b]
why: there are some codec problem between rtmpuser and rtmpuser(send and receive). I will fix it later( for handle flash to sip phone)
fixed(new direction): before stream callflow : web publish --RTMP-> red5 server --RTMP->red5phone server --RTP-> red5phone server --RTMP-> red5 server --RTMP -> web receiver
fixed stream callflow: web publish --RTMP-> red5 server--RTMP -> web receiver
[b]3. ONEWAY AUDIO in Transfer [/b]
why: can't exchange RTPStream 's remoteAddr correctly.
fixed: exchange RTPStream 's remoteAddr correctly. in the new direction(delete RTMP --> RTP --> RTP -->RTMP) just exchange their publish ID correctly.
音视频通话质量优化
本文主要探讨了音视频通话中常见的技术问题及其解决方案,包括反复尝试才能成功拨打电话的问题、RED5电话间的通话质量不佳及单向音频传输时的问题。通过对RTP流远程地址正确交换及调整通话流程等手段实现了问题的有效解决。
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