What is Outbound Proxy? Should I set an Outbound Proxy?
Outbound proxy is used to handle signaling and media traffic across a firewall, often in the presence of a security firewall/NAT entity. Generally, if you have an outbound proxy and you are not using STUN or other firewall/NAT traversal mechanisms, you can use it. However, if you are using STUN or other firewall/NAT traversal tools, do not use an outbound proxy simultaneously.
Can I configure SKYPE or IAX protocol on any Grandstream device?
No, unfortunately, both SKYPE and IAX use proprietary protocols, whereas all Grandstream devices adhere to the SIP protocol Version 2.0 RFC 3261.
What data range should I put for Layer 2 / Layer 3 QoS?
Layer 2 QoS – 802.1q VLAN 802.1p Priority
Since the VLAN tag is 12 bits, the value range is from 0 to 4095. 0 means no VLAN.
The priority value is 3 bits, so the value range is from 0 to 7.
Layer 3 QoS – DSCP
It has 6 bits ranging from 0 to 63.
What if my SIP URI domain is different from the SIP proxy server’s FQDN?
In Firmware 18.104.22.168 and later versions, you can place your SIP URI domain in the SIP Server field and the actual SIP server FQDN in the Outbound Proxy field. The phone will use the domain from the SIP Server as part of the SIP URI but will send and receive SIP messages through the SIP proxy server defined in the Outbound Proxy field.
What is Early Dial? Should I use it?
When you dial a number and do not press the “Dial” (“Redial”) or “#” button at the end of the dialed sequence, which is configured as the “Send” button, the phone sends the actual INVITE message about 4 seconds before the timeout. Enabling “Early Dial” will make the phone attempt to send INVITE messages on each keypress using the entered dialing sequence up to that point. If the SIP server supports a 484 Address Incomplete response, the phone will continue trying new keypresses until the entire entered dialing sequence is collected. This essentially eliminates the 4-second waiting time mentioned above.
Please note that this option should ONLY be used if your SIP server supports the 484 Address Incomplete response. Otherwise, other negative responses from the SIP server (such as 404 Not Found) will cause the call to terminate immediately.
What is STUN? Should I use it?
STUN stands for Simple Traversal of UDP over NAT. It’s a protocol that allows an IP phone to detect the presence and type of NAT behind which the phone is located. A STUN-capable IP phone, through a series of STUN queries to a STUN server located on the public Internet, can cleverly change the private IP address and port in the SIP/SDP message using the NAT-mapped public IP address and port it gathers. This allows the SIP signal and RTP media to successfully pass through a NAT without requiring any configuration changes on the NAT.
STUN provides a solution that works for most NATs that aren’t symmetric; for instance, most SOHO routers have non-symmetric NAT and in this case, using STUN is appropriate. However, STUN does NOT work with symmetric NAT, and if your routers have built-in symmetric NAT, do not use STUN.
Note: NOT ALL SIP PROXY SERVERS WILL WORK WITH STUN-MANGLING SIP MESSAGES, PLEASE CONSULT YOUR SERVICE PROVIDER FOR DETAILED INFORMATION.
What’s the difference between “User ID” and “Authentication ID”?
User ID is the user portion of the SIP address of the phone and is typically what’s displayed on the LCD as the Caller ID. For instance, it’s often a phone number or extension or a user’s name. Authentication ID is an ID that is used strictly for authentication purposes when the phone tries to establish a connection with the SIP server. It may or may not be the same as the User ID.
What number should I use for “TX Frames Per Packet”?
The choice depends on the codec component you select and the balance between bandwidth usage and the impact of packet loss. The larger the value, the higher the bandwidth usage, as more voice frames are packed into a UDP/RTP packet payload, thus the overhead of the network headers will be lower. However, the impact of a packet loss on perceived voice quality will be larger.
For PCMU/PCMA, the default is 2, and the maximum is 10.
For G723, the default is 1, and the maximum is 32.
For G726-32, the default is 2, and the maximum is 20.
For G729, the default is 2, and the maximum is 64.
For G728, the default is 4, and the maximum is 64.
Which NTP server can I use?
By default, the NTP server is set to “time.nist.gov” or “us.pool.ntp.org.” If either of these or your own NTP server is not working, you can try selecting an NTP server from the following link: http://www.eecis.udel.edu/~mills/ntp/servers.html or find more information at www.ntp.org.