When I applied the option which is " rtp.behindnat.send=true", Test result is good. I want to know whether packets is discarded in twenty more rtp session relayed at brekeke (include video/audio). How can I get log at brekeke to detect whether packet is dropped or not in relaying? because it is ...

Thanks for your reply.

I applied the option which is "net.rtp.udp.recv.bufsize=80000" you told me but I am afraid the issue might not be cleared.

I sent the captured file and result of test.

thanks,

I sent the captured file to your email.

I will check when the drop happen and tell you.

If I disable the option,
Does it lack as much performance compared to use TRUE option in relaying rtp packets ?

it's working well but the brekeke still drops 2~3 video RTP Packets for 5 minute in relaying. how does the configuration("net.rtp.native=false") affect the brekeke? I don't found it in administrator guide. Specific SIP UA sent rtcp-bye meesage to other side UA after 15minute in relaying between two ...

I have set the size to 80000, but it is of no effect. I think the brekeke drops also the audio packets. Max size of the audio packet is 160 bytes. I analyzed the packet dump captured with wireshark in sip server machine. I noticed something strange in the analysis. Although number of packets ...

I have already added the setting like below.
net.rtp.video.size = 8000

I will change the size value and try again.
I need your help about that some audio packet was droped in relay.

1. Brekeke Product Name and Version: Brekeke SIP Server Advanced version 3.4.6.3/411 2. Java version: 1.8.0_40 3. OS type and the version: Windows 2008 R2 Standard 4. UA (phone), gateway or other hardware/software involved: SIP UA (tx : 1200kbps/30fps /max frame size is 52400bytes) 5. Your problem ...

Log result is below. What is it mean? rtp.10006: close: total.pkt=0 packet/sec=0 rtp.10004: close: total.pkt=2 packet/sec=0 rtp.10002: close: total.pkt=1271 packet/sec=84 rtp.10000: close: total.pkt=622 packet/sec=41 rtp.10014: close: total.pkt=0 packet/sec=0 rtp.10012: close: total.pkt=1 packet/sec ...

the authorization header have not got username's value since version of xs20 was updated to 7
I will guery to CISCO.
thanks,

1. Brekeke Product Name and Version: SIP Server advanced Version 3 2. Java version: 64bit java 7 3. OS type and the version: windows server 2008 R2 64bit 4. UA (phone), gateway or other hardware/software involved: XS20(cisco) V7 5. Your problem: When xs20 send REGISTER Request to SIP Server, xs20 ...

when conference server received SDP from tandberg(sx20), conference server transmit RTP Video stream to tandberg thru SIP Server(RTP relay). SIP Server have received all rtp video packets from conference server but SIP Server droped some RTP video packets(5~10%) of all packets received when SIP ...

Even if I increase video buffer size to big(80000Byte), it have no effect on reducing packet drop. Sip server version is a 3.3.4.4 advanced in my customer site. Is there any other rule to drop RTP packet in SIP Server? SDP below o=tandberg 21 2 IN IP4 150.100.1.204 s=- c=IN IP4 150.50.59.21 b=AS ...

Increasing buffer size have a effect on reducing packet drop in version 3.3xxx. but it have no effect in version 2.4.8.6 stadard edition. here is SDP o=4002 0 0 IN IP4 192.168.1.30 s=UA Call c=IN IP4 192.168.1.30 b=AS:1500 t=0 0 m=audio 20098 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone ...

1. Brekeke Product Name and Version: SIP Server 3.3.4.4/379 2. Java version: 1.8.0_05 3. OS type and the version: Windows Server 2008 R2, 6.1 4. UA (phone), gateway or other hardware/software involved: TANDBERG C20 5. Your problem: SIP Server started on RTP relay option and local UA(A) send H.264 ...