Problems with video calls

mberg · Brekeke Junior Member Joined: 26 Oct 2007 Posts: 5

1. Brekeke Product Name and version:
Sip Server Version 2.2.1.6 Standard
2. Java version:

3. OS type and the version:
windows 2003 server

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:
Hi

I have made a program that can make video calls, all works fine when the calls are made from locations not behind NAT. But when the person is calling behind NAT and the calls gets RTP relayed by Brekeke Sip Server i only get 3-4 packet/sec from video rtp which is terrible. Which you can se below:

media video
transport RTP/AVP
payload 104
status active
listen-port 10006
send-port 10002
target 81.191.xxx.xxx:38062
packet-count 3030
packet/sec 3
current size 1312
buffer size 26000

media audio
transport RTP/AVP
payload 110
status active
listen-port 10004
send-port 10000
target 81.191.xxx.xxx:38060
packet-count 37269
packet/sec 47
current size 72
buffer size 260

I have tried to change the buffer size, with no luck.

Does anyone what causes this? Audio works fine.
The server has a 1000 Mbits connection, so that is not the problem.

Any help is appreciated.

Mberg

james · Brekeke Master Guru Joined: 10 Dec 2007 Posts: 501

Hi mberg,

>> I have tried to change the buffer size, with no luck.

How did you change the buffer size?

For Video, SIP Server's default RTP packet size is 2600 byte.
You can change it with "net.rtp.video.size".

If you want to set 3000 as a buffer size for example...
Go to [Configuration] -> [Advanced] page.
Add "net.rtp.video.size = 3000" there.
Then, restart the SIP Server.

mberg · Brekeke Junior Member Joined: 26 Oct 2007 Posts: 5

Hi

I have set the buffer size to 26000 byte, it had no affect on the number of packet/sec. I still only 3 to 4 packet a second. Is there a setting in Brekeke Sip Server that can be doing this?

Regards

Mberg

lakeview · Posted: Tue Oct 14, 2008 10:39 am Post subject:

If you use another Video client (X-Lite?) behind NAT, does the same issue happen?

mberg · Brekeke Junior Member Joined: 26 Oct 2007 Posts: 5

Hi

I have tried x-lite now and it works, i get about 30 packet/sec when BSS video rtp relay calls. It seems the problem is the softphone i am using (Eyeball Messenger 7.5). it limite the data sent from the caller, when BSS rtp relay the call and i don't know why.

Has anyone used Eyeball software with BSS before?

lakeview · Posted: Wed Oct 15, 2008 10:42 am Post subject:

Can you tune video codec type at the setting of Eyeball ?
Try another video codec.

mberg · Brekeke Junior Member Joined: 26 Oct 2007 Posts: 5

I changed the video codec in Eyeball to H263 and BSS informs me that the rtp video payload is h263, but still i only get 3-4 packet/sec. current size on packet can be all from 20 up to 1312 byte, this changes every second. I will try to get some answers from Eyeball.

audio rtp is stabile at around 50 packet/sec at packet size 72 byte.

Do not understand why RTP handles diffrently when using Eyeball and not X-Lite.

lakeview · Posted: Wed Oct 15, 2008 1:48 pm Post subject:

I think.. Eyeball doesn't send video RTP packets to the SIP Server correctly.

Do you have any packet-capturing software? (Wireshark for example.)

Capture packets during a video call. and then check the video RTP's destination port number.
The port number should be the same number [listen-port] indicates .