Some RTP packets will be dropped because they either arrive before the signalling starts the media (on 18x or 200), or they arrive after the hangup. How long was the call you made? 2380 suggests under a minute? Try a much longer call and see whether you're still seeing the same amount of loss. -Steve On 18 March 2011 07:21, joy this <thisjoy0528@gmail.com> wrote:
I have some tests for the problem that the voice keeps off and on during the call. I use Wireshark to check the package flow on the caller, callee, and FS sides. The network structure: caller(PCMA)---(Wi-Fi)--->WAN--->FS--->callee(PCMA). FS and the callee are under the same router. The result is: The caller sends 2688 packages to FS, and FS receives 2687 packages. Then FS sends only 2380 packages to the callee, and the callee receives 2380 packages. There is about 11.4% package loss in FS, and the callee hears incomplete voice.
I have some questions:
1. I guess FS drops packages from the caller because these packages arrived too late. Is it reasonable? If not, what is the reason?
2. I want to check if FS drops packages because of the late arrival. Could you tell me the packages flow in the source code?
3. I set the jitterbuffer_msec from 10 to 10000, but it seems not effective. Wouldn’t the jitterbuffer_msec affect the result?
Sincerely yours,
thisjoy.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org