GRANDSTREAM NETWORKS

Transcription

1 GRANDSTREAM NETWORKS FAQ - Frequently Asked Questions Grandstream Networks devices specific questions 1. Are BudgeTone SIP phones certified for use in North America and Europe? 2. What are the differences among BudgeTone-101, 102 and 102D models? 3. Can I call another IP phone using its IP address directly without a proxy? 4. Where to get the latest software release my Grandstream SIP devices? 5. How do I upgrade my Grandstream's firmware? 6. How do I setup my tftp server on the phone? 7. Does Grandsteram Network Inc have a tftp server for me to use to upgrade my Grandstream BT100 series phone? 8. How do I know which firmware version my Grandstream phone is running? 9. Why is my phone's LCD keep lighting up without showing any date? 10. Why is my phone showing date " "? 11. Which NTP server can I use? 12. How to reset my Budgetone 100 phone to factory default setting? 13. Do you have other colors than the white one? General SIP and VoIP FAQ 1. What is Outbound proxy? Should I put an Outbound proxy in the field? 2. What is the difference between "User ID" and "Authentication ID"? 3. What if my SIP URI domain is different from the SIP proxy server FQDN (Fully Qualified Domain Name)? 4. What types of voice codec do Grandstream VoIP products support? 5. What Codec should I use for my Granstream phone? 6. What is "Voice_Frames_Per_TX" and how does it relate to Ethernet traffics? 7. What number should I use for "Voice Frames per TX"? 8. How much overhead does Ethernet add to RTP packets? 9. What is the frame rate and bit rate for each codec? 10. Why is G.723 the best option for narrow bandwidth IP communication? 11. What is??early Dial?, Should I use it? 12. What is STUN?, Should I use STUN? 13. Do I still need to put in "Outbound" proxy if my phone is working under STUN? 14. Which other 3rd party SIP applications and products are BudgeTone SIP phones compatible with? 15. Which SIP based IP telephone service providers do you currently support? Service providers configuration settings 1. How do I setup my Grandstream Phone for Delta3/iconnect network? 2. How do I setup my Grandstream Phone for nikotel network? 3. How do I setup my Grandstream Phone for MCI(test) network? 4. How do I setup my Grandstream Phone for telic.net network? 5. How do I setup my Grandstream Phone for go2call network? 6. How do I setup my Grandstream Phone for FWD service?

2 Grandstream Networks devices specific questions 1. Are BudgeTone SIP phones certified for use in North America and Europe? Yes. The BudgeTone-101 and BudgeTone-102 IP phones are FCC (part 15) and CE certified and thus can be used in the US and European market. Its universal power adaptors are UL certified. 2. What are the differences among BudgeTone-101, 102 and 102D models? Within the BudgeTone-100 series family, model 101 and 102 have the same software functions and the only difference is that model 101 has 1 (one) Ethernet interface and model 102 has 2 (two) Ethernet interfaces. Model 102D will have a better character based LCD, 2 (two) Ethernet interfaces, and more software functions such as 3-way conferencing, support for SIMPLE, support for more vocoders, future support for power-over-ethernet, ear phone interface, etc. 3. Can I call another IP phone using its IP address directly without a proxy? Yes. Direct IP-to-IP calling is supported. Please refer to Users Manual for details. 4. Where to get the latest software release for Grandstram SIP devices? Please visit: for detail. 5. How do I upgrade my Grandstream's firmware? First configure your tftp server on the phone and then power cycle or reboot the phone. Please consult user manual for detail. 6. How do I setup my tftp server on the phone? There are 2 ways setting up your tftp server. 1) From the phone's web page. 2) From phone keypad, press menu button and down arrow to item number 6, press menu button one more time to get into the "Editing" mode. If the tftp server IP displayed is not the one you want, enter the entire 12 digits of the IP address of your tftp server. e.g., if your tftp server is , enter Does Grandsteram Network Inc have a tftp server for me to use to upgrade my Grandstream device? Yes. Grandsteam provides a NAT-friendly high-performance tftp server for its customers. The IP address of this tftp server is listed under the "Services" section of its Web site at: 8. How do I know which firmware version my Grandstream phone is running? There are two ways that you can see your phone's firmware version: 1) use "menu" button. Press "menu" and then up arrow button 3 times to get to item number 9, "code rel", and then press "menu" one more time to get into the "code rel" checking mode, use up or down arrow button to see "B (bootloader)" and "P (program)" code date and version. 2) From Grandstream phone's web page.

3 The software version should be displayed on top of the web configuration page in a format similar to the following: Software Version: Program--1.0.X.X Bootloader--1.0.X.X HTML--1.0.X.X 9. Why is my phone's LCD keep lighting up without showing any date? The most possible problem will be that the phone is not getting responses from the NTP server. Check your network connection, DNS server or try to use another NTP server. 10. Why is my phone showing date " "? This is probably because either DNS is not resolving the NTP server correctly, or the NTP server is not responding. This is the default date shown under this circumstance. 11. Which NTP server can I use? We use "time.nist.gov" as default NTP server. If that or your own NTP server does not work, try to select an NTP server from following link: Or you can get more info from How to reset my Budgetone 100 phone to factory default setting? 1. set up the phone to have NO tftp server and then boot without network 2. Press the Menu and then the upward arrow key, you will see the Reset option on the LCD. At this point, enter the full 12 digit of the MAC address or Product ID of your phone (printed at the back of your phone). e.g., for "A/B/C", press the "2" button until you see "A" or "B" or "C" appears, for "D/E/F", press the "3" button until you see the corresponding character. Then continue to enter the complete 12 digit of the MAC address. After that, press "Menu" again. 13. Do you have other colors than the white one? We will have another dark gray color option in the near future. General SIP and VoIP FAQ 1. What is Outbound proxy? Should I put an Outbound proxy in the field? An Outbound proxy is mostly used in presence of a firewall/nat to handle the signaling and media traffic across the firewall. 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/mat traversal tools, do not use an outbound proxy at the same time. 2. What is the difference between "User ID" and "Authentication ID"? User ID is the user part of the SIP address of the phone and this is usually the information displayed as Caller ID on the LCD. e.g., typically it is a phone number or extension number or a user's name. Authentication ID is an ID used strictly for authentication purpose when the phone attempts to contact the SIP server. This may or may not be the same as User ID. 3. What if my SIP URI domain is different from the SIP proxy server FQDN (Fully Qualified Domain Name)? With firmware and later, you can put the your SIP URI domain name into the SIP Server field, and put the actual sip server FQDN into Outbound Proxy field. The phone will use the domain name in SIP Server as part of SIP URI but send and receive SIP messages through the SIP proxy server defined in the Outbound Proxy field. 4. What types of voice codec do Grandstream VoIP products support? Each codec has its uniqueness for certain application. Grandstream BudgeTone series VoIP phones and HandyTone series analog telephone adaptors support G.711-uLaw, G.711-aLaw,

4 G.722, G.723, G.726, G.728 and G.729 and ilbc. The wideband codec G.722 is only supported by BudgeTone VoIP phones. It has the same bit rate as G.711 but with twice sampling rate (16KHz vs. 8KHz) and better sound effect. 5. What Codec should I use for my Granstream phone? Generally speaking, all codecs provide good voice quality. However, lower bit rate codec may have poor quality for music. DTMF tones and fax signals on audio channel may not be decoded on remote premises. If the bandwidth allows, G.711 is the default option, G.722 give even better sound quality. By default, PCMU(G711u) will be used. Both PCMU and PCMA will give you toll quality but their bandwidth consumption is also the highest (64kbps). If your network bandwidth is low, you can choose other lower-bit-rate codec such as G723 or G729 which will give you near toll quality at much smaller bandwidth consumption (G723 consumes 5.3/6.3kbps and G729 consumes 8kbps). If bandwidth is not a concern and you want good voice quality, try using PCMU or PCMA, or even the new wide band codec G722 (64kbps) which will provide hi-fidelity voice that is better than toll quality. 6. What is "Voice_Frames_Per_TX" and how does it relate to Ethernet traffics? To reduce the overall Ethernet/IP/RTP overhead introduced by the 54 bytes header, multiple voice frames can be packed into single Ethernet frame to transmit. Of course, this would increase the voice delay. In case that network bandwidth is constrained, increasing this count may improve the overall voice quality. If RTP packets are sent every 2.5ms (for G.728), the total Ethernet/IP/RTP overhead is 0.432*400 = 172.8kbps. This won't work well over public Internet. However, if RTP packets are sent every 10ms, the total Ethernet/IP/RTP overhead is down to 0.432*100=43.2kbps. If RTP packets are sent every 20ms, the total Ethernet overhead may be further down to 0.432*50=21.6kbps. We suggest 30ms packet rate for G.723/iLBC and 20ms for the rest codecs. Voice_Frames_Per_TX is then set to 1 for G.723/iLBC, 8 for G.728 and 2 for all the rest. 7. What number should I use for "Voice Frames per TX"? It depends on what codec you choose and balance between bandwidth utilization and impact of packet loss. The bigger this value, the higher bandwidth utilization because more voice frames are packed into the payload field of a UDP/RTP packet and thus the network header overhead would be lower. However, the impact of a packet loss on perceived voice quality will be bigger. For PCMU/PCMA, the default is 2 and max is 10 For G723, the default is 1 and max is 32 For G726-32, the default is 2 and max is 20 For G729, the default is 2 and max is 64 For G728, the default is 4 and max is How much overhead does Ethernet add to RTP packets? For voice over IP over Ethenet, an RTP packet contains 54 bytes (or 432 bits) header. These 54 bytes consist of 14 bytes Ethernet header, 20 bytes IP header, 8 bytes UDP header and 12 bytes RTP header.

5 9. What is the frame rate and bit rate for each codec? G.711 has 10ms frame length with 64kbps bit rate; G.722 has 10ms frame length with 64kbps bit rate; G (also referred as G.721) has 10ms frame length with 32kbps bit rate; G.728 has 2.5ms frame length with 16kbps bit rate; G.729 has 10ms frame length with 10kbps bit rate; G.723 has 30ms frame length with either 5.3kbps or 6.4kbps bit rate. ilbc has 20ms or 30ms frame length with 15.2kbps or 13.3kbps bit rate; 10. Why is G.723 the best option for narrow bandwidth IP communication? For G.723, the frame rate is at 30ms and the codec bit rate is 5.3kbps (20 bytes per 30ms) or 6.4kbps (24 bytes per 30ms). The total bit rate are *33.3 = 19.7kbps, or *33.3 = 20.8kbps. This low bit rate is ideal to transmit over 28.8kbps dialup modem connection. With other technologies like data link layer compression, silence suppression and comfortable noise generation, the total bandwidth could be even further lowered. 11. What is??early Dial?, Should I use it? When you dial a number, if you do not press the "Dial" ( "Redial") or "#" key if it is configured to function as the "Send" key at the end of your dialed string, the phone will wait for about 4 seconds before timeout and then sends the actual INVITE message. If you set "Early Dial" to be YES, then the phone will attempt to send out INVITE at each key input using the entered dial string collected so far. If the SIP server supports 484 Incomplete Address response, the phone will keep trying with each new key entry until the complete dialed string is entered. This will essentially eliminate the 4-second wait time mentioned above. Please note that this option can be used ONLY when the SIP server supports 484 Incomplete Address response. Otherwise, any other negative responses from the SIP server (such as 404 Not Found) will cause immediate termination of the call. 12. What is STUN?, Should I use STUN? STUN stands for Simple Traversal of UDP over NAT. It is a protocol which enables an IP phone to detect the presence and type of NAT behind which the phone is placed. An IP phone that supports STUN can intelligently modify the private IP address and port in its SIP/SDP message by using the NAT mapped public IP address and port through a series of STUN queries against a STUN server located on the public Internet. This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT. STUN presents a working solution for most NATs that are not symmetric NAT, e.g., most of the SOHO routers have non-symmatric NAT and in this case, it is OK to use STUN. 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 SERVER WILL WORK with A STUN TRANSLATED SIP MESSAGES, PLEASE CONSULT YOUR SERVICE PROVIDER FOR DETAIL.

6 13. Do I still need to put in "Outbound" proxy if my phone is working under STUN? NO. 14. Which other 3rd party SIP applications and products are BudgeTone SIP phones compatible with? We have been active participants in the SIPit events and have done extensive tests successfully with a number of 3rd party SIP products directly or indirectly through our customers or partners. A partial list of other products with which we have successfully tested basic and in some cases advanced features include: Cisco (7960/7905 IP phones, ATA186, SIP proxy, 5300/3640, etc) Microsoft (Messenger, RTC Server) DynamicSoft (SIP proxy) Broadsoft (softswitch) Santera/Tekelec (softswitch) Siemens (IP phone) Nortel (softswitch, softphone) Intel (gateway) Lucent (media server) Alcatel Jasomi (border controller) iptel/ser (open source SIP proxy) Digium/Asterisk (open source IP PBX) vovida.org/vocal (open source IP PBX) Mitel (IP phone) Pingtel (IP phone, softphone) Teledex (IP phone) Dlink (IP phone) Ingate/Intertex (SIP aware firewall) Hotsip (SIP proxy, STUN server) Radvision Sylantro Hughes Software Systems Avaya Ericsson Nokia Sharp TI/Telogy edial (conferencing server) octave (conferencing server) Cosmocom (contact center application) IVR Technologies (software based IVR system) SoftFront (SIP proxy, UA) Vegastream (gateway) UTStarcom (SIP proxy, media gateway) Tangerine (SIP proxy/registrar) and many others Which SIP based IP telephone service providers do you currently support? We have tested with and support the following service providers' SIP network:

7 MCI Nikotel Delta3 Telic.net Go2Call Free World Dialup This list will continue to expand and please check for updates from time to time Service providers configuration settings 1. How do I setup my Grandstream Phone for Delta3/iconnect network? SIP Server: natrelay.deltathree.com outbound proxy: leave it blank User ID: xxxxxx (your Delta3 account number) Authentication/Login ID: xxxxx (same as above, your Delta3 account number) Password: xxxxx (your Delta3 password) Dial plan: How do I setup my Grandstream Phone for nikotel network? SIP Server: calamar0.nikotel.com Outbound proxy: leave it blank User ID: xxxxx (your nikotel account number) Authentication ID: same as your User ID Password: your nikotel password NAT Traversal: YES (WITHOUT setting the STUN server) 3. How do I setup my Grandstream Phone for MCI(test) network? SIP Server: siptest.mci.com Outbound proxy: (use an outbound proxy if MCI provides one for you) User ID: xxxxx (your MCI assigned account/phone number) Authentication ID: (Your MCI assigned id, i.e., foo) Password: your MCI password

8 NAT Traversal: No (You need to set up your STUN server if you don't have outbound proxy) Note: MCI Proxy server seems to respond our phone client SIP messages correctly. 4. How do I setup my Grandstream Phone for telic.net network? SIP Server: sip.telic.net Outbound proxy: (Use outbound proxy, it will not work under STUN for now) User ID: xxxxx (your Telic.net account number) Authentication ID: same as your User ID Password: your Telic.net password Note: STUN is not working yet against Telic.net's SIP proxy server for now. 5. How do I setup my Grandstream Phone for go2call network? SIP Server: voip01.go2call.com Outbound proxy: (Should leave it blank, because it's a GW) User ID: xxxxx (your Go2Call PIN number) Authentication ID: same as your User ID Password: xxxxxxx (Your Go2Call password) NAT Traversal: YES (WITHOUT setting the STUN server) 6. How do I setup my Grandstream Phone for FWD service? SIP Server: fwd.pulver.com outbound proxy: :5082 (used only when behind firewall, otherwise leave it blank) User ID: xxxxxx (your FWD account number) Authentication/Login ID: xxxxx (same as above, your FWD account number) Password: xxxxx (your FWD password) NAT Traversal: No (You need to set up your STUN server if you don't have outbound

9 proxy)