[Freeswitch-users] RTP Loss with Freeswitch in Default Mode
vikas452 at gmail.com
Wed Oct 11 19:33:13 UTC 2017
Thanks Michael for responding.I will check that.
I also found difference in the RTP packet count received on A Leg port of
Freeswitch and sent out from B Leg port.I have taken packet dump on server
and checked this in RTP streams stats in wireshark.
For example 500 video packets were received on A Leg of Freeswitch and then
70 packets were sent out from B Leg to the private ip/port and 350 were
sent to public ip/port but still 80 packets are missing.I found a
significant difference for bad network condition.
Could you please give me some suggestions for this issue??
On 12-Oct-2017 12:07 AM, "Michael Jerris" <mike at jerris.com> wrote:
check out the confluence page on handling nat, and make sure your clients
are sending the public not the private ip for media addr.
On Oct 11, 2017, at 6:26 AM, vikas sharma <vikas452 at gmail.com> wrote:
I have also found that Freeswitch starts sending RTP on the private ip/port
first.Ideally it shall not be there.Please let me know if there is any
specific setting for RTP nat handling so that Freeswitch send directly on
public ip/port only.
On Wed, Oct 11, 2017 at 12:03 PM, vikas sharma <vikas452 at gmail.com> wrote:
> Can i get some suggestions to improve the video call quality and reduce
> packet loss with Freeswitch server.
> On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma <vikas452 at gmail.com> wrote:
>> Anything that can help me in this regard??
>> On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma <vikas452 at gmail.com> wrote:
>>> I am making a linphone to linphone call using Freeswitch server in
>>> Defualt mode but i am facing huge RTP loss and the video call is getting
>>> stuck in between.I have also found that there is a difference between the
>>> RTP packets received on A Leg and sent from B-leg and vice versa.
>>> Can somebody help me to understand what exactly happens with the RTP
>>> packets on server and why there is a difference in the RTP count??
>>> How can we improve the video call quality in moderate network conditions.
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