I'm a newbie software developer who develops SIP/RTP Voip software. For sure, I am using UDP protocol and Video Codec for this video is H264.
Since I am new to this Voip area, I am so confused and suffering painful network issues a lot.
I would like to ask experts something related to Network specifically dealing with RTP/RTCP issues on Jitter / Packet loss.
After SIP successfully creates media session, I get some issue for QoS.
Problems I am facing up is just like this below.
Wifi network(latency : 11.1m/s download speed : 14.9mbps upload speed : 3.27mbps) :
http://www.youtube.com/watch?v=epm01c6IT5Q&feature=youtu.be
3G network(latency : 26.4M/s Download speed : 1.94Mbps upload speed : 2.42Mbps) : http://www.youtube.com/watch?v=-iG156_wdQE&feature=youtu.be
as you see, through 3G that has low upload and download and unstable latency, video quality including video issue coloured green and video's delay is better than Wifi.
using 3G network slower than Wifi , I can always take better user experience than Wifi.
I didn't analyse RTP/RTCP packets deeply but the thing I can tell is ...
At the problem situation, when Wi-fi was used for the application, Jitter was strangely high enough and packet loss was obviously high as well.
To sum up,
- As you can see, video quality is better when I use 3G network slower than Wifi.
- When Wifi works there, Jitter and packet loss are obviously high as I can analyse packet using wire-shark on the receiver side.
- at that morning time, video problem(green pixel of video, video delay) was much more serious but as time went, at the afternoon and night, problem had been recovered a bit.
As far as I know, it would be related to network bandwidth and network congestion. I am not sure that it is proper diagnosis and also need to solution to this. I'm sorry that I have not enough background information yet.
Thanks.