no video relay

CastB · Post by **CastB** » Sat Feb 05, 2011 7:20 am

1. Brekeke Product Name and version: Sip Server 2.4.7.3/286.1

2. Java version: 1.6.0_17

3. OS type and the version: 2.6.18-194.32.1.el5.centos.plus

4. UA (phone), gateway or other hardware/software involved:
X-lite on windows7

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Sip Server is in the public and both softclients are in the same LAN behind NAT

6. Your problem:

Hi,

We are currently testing the relay function from sip server. Audio is relayed, however there is no video. We already tried other softphones and other codecs with same result. We also set the video buffer size to 6000, no result either.

We also did run a wireshark capture and it seems that video packets are being send to the sipserver but packetcount in the session details page states 0 (zero)

Session details
EX-SID 85
From-uri sip:e_4@MySipServerFqdn [behind NAT]
From-ip MyNatGlobalIP:63800 (UDP)
From-if MySipServerIP:5060
To-uri sip:e_5@MySipServerIP:5060 [behind NAT]
To-ip MyNatGlobalIP:49488 (UDP)
To-if MySipServerIP:5060
Call-ID NjA3ZjUwZjg0YTUyMWMzNzQ4YTY3NmY5YTRhY2Y4MjI.
rule registered=sip:e_5(sip:e_5@MyNatGlobalIP:49488)
plug-in InviteSession
sip-packet-total 11
listen-port 5060
session-status Talking
time-inviting Sat Feb 05 10:41:59 CET 2011
time-talking Sat Feb 05 10:42:09 CET 2011
length-talking 00:00:40
rtp-relay on
rtp-srcdst
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10012
send-port 10008
target MyNatGlobalIP:55258
packet-count 1626
packet/sec 36
current size 172
buffer size 260
media video
transport RTP/AVP
payload -
status active
listen-port 10014
send-port 10010
target MyNatGlobalIP:51992
packet-count 0
packet/sec 0
current size 0
buffer size 6000
rtp-dstsrc
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10008
send-port 10012
target MyNatGlobalIP:50362
packet-count 2157
packet/sec 43
current size 172
buffer size 260
media video
transport RTP/AVP
payload -
status active
listen-port 10010
send-port 10014
target MyNatGlobalIP:49532
packet-count 0
packet/sec 0
current size 0
buffer size 6000

Wireshark details
Stun binding succesfull request for <MyNatGlobalIP> ports: 50362,50363,49532,49533

Audio from <MySipServerIP> to <MyLocalIP>
RTP PT=ITU-T G.711 PCMU, SSRC=0xC4B7F636, Seq=5825, Time=833120
User Datagram Protocol, Src Port: 10012 (10012), Dst Port: 50362 (50362)

Audio from <MyLocalIP> to <MySipServerIP>
RTP PT=ITU-T G.711 PCMU, SSRC=0x991FB3EA, Seq=468, Time=1772860
User Datagram Protocol, Src Port: 50362 (50362), Dst Port: 10012 (10012)

Video from <MyLocalIP> to <MySipServerIP>
H.263 RFC4629 PT=DynamicRTP-Type-115, SSRC=0x6BAE38DC, Seq=6631, Time=333627 (PSC)
User Datagram Protocol, Src Port: 49532 (49532), Dst Port: 10014 (10014)

So it seems video is being sent to the Sip Server...

Anyone any hints how to solve this?
Thanks, CastB

hope · Post by **hope** » Mon Feb 07, 2011 12:04 pm

at brekeke sip server /configuration/RTP/
maybe need to set port for video. default value are 0.

janP · Post by **janP** » Mon Feb 07, 2011 6:04 pm

hope,
The video port can be 0. you don't have to tune it.
The admin document said "If set to “0”, the server uses the same port range as Audio".

janP · Post by **janP** » Mon Feb 07, 2011 6:08 pm

CastB,
If you enable the "RTP-Relay" without STUN, does the same problem still happen?

I mean..
Put both video clients and SIP Server on the same local network (without NAT..).
Set the "RTP relay = ON" at the [Configuration] > [RTP] page.
Make a video call through the SIP Server between video clients..

If so, does the SIP Server relay video packets correctly???

CastB · Post by **CastB** » Tue Feb 08, 2011 5:46 am

Hi,

Thanks for the tips.

The video ports where set to 0, bu just to be sure i tried adjusting the video port range but that indeed did not solve the problem.

I will try without STUN in the same LAN asap, get back to you on that.

taitan · Post by **taitan** » Tue Feb 08, 2011 6:02 am

Depending on NAT type and sip client, stun may not work as you want...

Disabling of stun at sip client will solve the problem.

The sip server can follow the video port automatically if stun is disabled.

CastB · Post by **CastB** » Mon Feb 14, 2011 9:35 am

Hi all,

Thanks for all the tips. We solved the problem by debugging it in a local network configuration.

The problem was in the "connection tracking" module we used in the firewall. (kernel module ip_conntrack_sip). Video packages were not correctly passed through.

We already tried it outside the LAN and that seems to work now also.

janP · Post by **janP** » Mon Feb 14, 2011 11:56 am

glad to know it