PLANET VIP-350PT - Manual (Page 87)

[87/168]
• 000 - Routine (normal)
2. Bit 4: 0 = Normal Delay, 1 = Low Delay (RTP/RTCP default) .
3. Bit 3: 0 = Normal Throughput, 1 = High Throughput (RTP default).
4. Bit 2: 0 = Normal Reliability, 1 = High Reliability (RTCP default) .
5. Bit 0-1: Reserved for Future Use.
System default is 0xb8 (184 in decimal) for RTP and 0x74 (116 in decimal) for RTCP packets. Please
enter it in decimal.
I
P T y p e - o f - S e r v i c e
[ 0 - 2 5 5 ]
1 8 4
Del Back
Symmetric RTP Flow
The IP Phone supports symmetric RTP flow for the cases where only one endpoint is behind a
NAT, and RTP packet flow will be possible in at least one direction. Since a client behind a NAT
can usually successfully send RTP packets to another client in the public Internet, in a symmetric
mode, RTP sent in the other direction could be sent to the address and port that RTP was
received from. As a result, even if you are behind a NAT and choose “Full Access” as your way
to traverse NAT, you could still reach another IP Phone on the public internet (since the peer
would adjust RTP flow accordingly).
Generally, if your terminal is behind a symmetric NAT, which cannot be traversed by STUN, or
your terminal is NATed, you should activate this feature. And the other party on the WAN (or
capable of traversing the NAT it currently behind without problem) should
NOT enable this
feature.
Once “Symmetric RTP flow” is enabled, IP Phone will send its INVITE message stipulating its
desire for “Symmetric RTP flow” in SDP during call setup phase if the “UDP traversal” mode
configured in 『Main Menu』=>”6.Network” / ”NAT & firewall” / ”UDP Traversal” is either “Full
access” or “Enable STUN”. That is if you have configured “Static NAT map” as your way to
traverse NAT, it will not request the peer to enable symmetric RTP flow). The SDP sent by this
UAC would be:
o=SIP-Phone 28908445311 28908445311 IN IP4 192.168.3.51
s=-
t=0 0
c=IN IP4 192.168.3.51
m=audio 35000 RTP/AVP 4 18 0 8 101
a=rtpmap:101 telephone-event/8000
a=direction:
active IN IP4
If the UAS supports this extension (which VIP-350PT/ VIP-550PT does), it will wait for RTP packets to
be received from the client behind the NAT before sending, the answer SDP will be:
v=0
o=SIP-Phone 28908445405 28908445405 IN IP4 192.168.3.101
s=-
t=0 0