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voip not usefully - last hope hardware telephone?2010-09-07 | |||||||||||||||||
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I would like to replace the similar fixed mains connection for telephone calls on mobile phone + to fixed net through voip, have me therefore testweise with justvoip.com (cheaply however probably not so reliable) as well as sipcall.ch (the CH seems to be one the best? in) announced. In order to clarify, whether for me voip an alternative is actual, I tested the following variants with sipcall.ch: - Softphone on HTC dia. moon (WM6 standard voip and Portsip) -> Would like actually to me, but not useful (delay to 1s, much to quiet etc.) - Softphone on PC (X-Lite of Sipcall) with Headset -> Delay still over 1/4s and receiving station has greyful echo of itself. Now ask I, doesn't go themselves that really better? sipcall/phonestar “promise” fixed net quality etc. Exists with a hardware Tel. (e.g. Siemens Gigaset C470 IP) hope for improvement? I would have unfortunately to buy the part, in order to find out  Infrastructure: Rout: Linksys WRT54g with Thibor/HyperWRT firmware (QoS activated) Internet: Cablecom 15 ' 000 A Ping also - l 200 at the Sipcall Proxy results in 15-20ms Thank you for your pieces of advice!!!! X-Lite is actually already completely good. A hardware telephone will probably bring nothing. You can test times another Softphone. z. B. of Snom, that nearly exactly, as the hardware telephone of snom functions: http://www.netzwelt.de/download/4013-snom-360.html That could be done quickly, thanks!! OK ONE, i.e. thus, which is at least as good (good) a Softphone as a Hardwarephone? I wird's times with snom try! White however genuinly not which with me wrongly runs, can to me hardly to present someone, or even companies, then to telephone be able… I believe that also, because hardware telephones consists also of a computer with software, also still another much weaker processor, than a PC has. It could however a problem with yours routs gives or the PC-Firewall. You tried it already times on another computer. VoIP over UMTS is not recommended unfortunately momentarily because of the strong delay. At the PC or Harwaretelefon one has also a delay, which is hardly noticeable however. Quotation:
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Have completely forgotten to say: The HTC dia. moon was connected for the test by WLAN in the same network as the PC with Softphone! A PC-Softphone will however better and better be probably as one on the Smartphone, or? In order to ask times completely simply: How ist's usually at the best one (small Delay, no echo etc.)? - Call by call - hardware IP telephone - Softphone on PC - Softphone on PDA/Smartphone Thank you for your Tipps, I will not so fast give up!!! more @Voip_Indianer To know well that it works so well. Which program do you take in addition? @r1huber Henn you with the HTC by Wlan was connected, you probably the wrong program installed. Which HTC do you have? CCU, Windows version, free memory? Quotation:
The delays are already extremst nervig with VoIP of over DSL, but with VoIP of over UMTS it is still worse. Quotation:
Also at the input devices it can lie. With hardware VoIP telephones particularly on it co-ordinated microphones and loudspeaker are used. And so on… Simply times hardware VoIP equipment tries out, like e.g. snom 300. Greeting [quote=Tippfehler; 1488744] Which program do you take in addition? Quotation:
Behind the WBR3800 is a WLAN rout Linksys WRT54G with open WRT, which administers the internal LAN here. SIP of terminals are attached to it - by cable - the following: 1 * Siemens optipoint 410 1 * Cisco (Linksys) SPA 2102 Optipoint computing centre is connected to a rented server in the Strato, on which a Asterisk runs. There the different VoIP Provider is administered. The SPA2102 connects with a haven directly too sipcall. This connection is used for the fax equipment. The second haven of the SPA connects also to the Asterisk servers, at this haven is attached a cordless DECT telephone. Echoes and Delays are not present with VoIP discussions. The connection of the LAN in Internet is made over a VPN Routing to the server with Strato. That means, all Traffic from my LAN is geroutet first times by the VPN. From software solutions for SIP do not think I in principle much. One should not use it as yardstick for quality statements. more @Tippfehler: HTC dia. moon and/or. MDA Compact 4, 528Mhz, WM6.1, RAM freely approx. 20MB Be afraid, the Softphone variant on a PDA a Gewrge will be probably always… Will times the PC without rout to the Cablemodem to hang, wenn's then becomes not better remains only a IP telephone. For what hold you from the Gigaset C470 IP? @George99: Nervige delay in DSL?? Then wird's with me with Cable become better also never… Does someone have a notion, in which range (ms) the latency should lie with optimal configuration about? Quotation:
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http://wiki.ekiga.org/index.php/Windows_Users Is the only Softphone with echo cancel, which however because of the synchronisation mostly just functioned if Micro and loudspeakers are attached at the same audio chip. With you is the case? Thus with Lausprecher at the Klinkenbuchse however micro at the USB Headset does not go that mostly already any longer. Quotation:
Hello to all! In the following I gathered times my experiences about VoIP. My remarks are meant not only as answer to the places quoted of me, but rather as general statements about VoIP and its problems. Quotation:
I have before I VoIP so well to run had gotten, at the beginning also problems had. With test and the error tracing thereafter I stated then that it was appropriate for hardware, software, DSL connection or VoIP Provider always to “more wrongly” or “more unausgereifter”: Softphone of problems: Power (e) problems with some sound maps in the PC/laptop. -> There was then a time disalignment in the discussion Solution: Other PC/laptop no problem  Hardware problem of Nr.1 - which rout: Depending upon manufacturer (D-left, Netgear, T-Com) problems with the Firewall, rout made close despite release haven (? 1?) -> It rang, but one did not hear anything or the interlocutors heard one not, etc. Thus (was) the only solution would be with this routing a DMZ, which I found however for safety reasons a very bad solution! Solution: Rout from Intertex or Lancom. And already I had no more problems with discussion aborts and einseitingem hearing. Whereby I here still which to the support say must: That Support of Intertex top class is, this sent to me within 2 days a firmware co-ordinated with my DSL offerer, so that it comes to no more a choice difficulties! Super! That Support of Lancom is not so mad Perhaps it would be times useful, even if the coworkers the enamels and not only “standard responses” and “FAQ excerpts” would read correctly there would dispatch, which already admits to me were and whom do not meet core of the problem. (There a permanent condition seems unfortunately to be!) Which with this problem is however amusing, in the next beta or release firmware those are fixed the support of communicated problems or as a “acquaintance restrictions” defines…  “Hardware problem of Nr.2 - the SIP telephones” As the first I tried out a Grandstream GXP-2000: The calling heard me only very quietly and I had also my listener volume on full to place, so that I could hear reasonably. When acquitting the discussions were partly chopped off. Solution: SNOM! Property meanwhile various telephones of it in the employment and no more problems. Should nevertheless times which be, SNOM has supper support (usually detailierte answers (solutions) within 3 hours!) DSL line problems: The Alpha and Omega is with VoIP a fast and stabiele line! If one does not go tried on a DSL Light VoIP to make, that simply! Solution: One should possess a line starting from 2000er. I personally have 16000er with 15-18ms Ping time. VoIP Provider of problems: Depending upon used Provider one has more or less losses and problems. Which I determined thereby am, that the “Billigheimer” (Betamaxe, etc.) (much) more “disturbances” have, than “fernnftig calculating” offerers (Sipgate, Dus.net, etc.). Solution: Provider change Quotation:
As Resueme or summary of my experiences, I can only say that one, with the “suitable” hardware and the “correct” DSL and VoIP Providern, can telephone very well and in exzellenter language quality more ner to Internet. Greetings CCU professional I give you in many points quite, but that here: Quotation:
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It is very probably conscious to me that I with “worse” codecs like e.g. LPC10 a smaller range needs. One should consider then in addition, those seriously bad Ping (periods of reply) of the “line”, whereby then reasonable communication not is more possible. Because then we are again as with the voice transmission (one speaks and the other one waits). Listen to you nevertheless times the quality differences of the codecs: http://www.voipforo.com/en/codec/audi_codec.php And with these photographs the Ping (periods of reply) is not along (there is not present, there best laboratory conditions) It had thus its reason why I with the codecs was not concerned, because we here over the “quality and usefulness of VoIP” in Germany and/or. Europe and not in Africa Greetings CCU professional Quotation:
 By the way also I talk only about the voice communication. The mentioned VoIP conversion over analog modem very probably serves communication between developing countries and Germany. And despite sufficient range the available I work here in Germany excluding narrow-band with G729 - without so far somebody had weighted because of bad language quality. Quotation:
: One measures the quality/quality of a codec in the MOS value. The more highly the value (max. 5 is, min. is 1) all the better (more originally, more in a true-to-life manner and more naturally) is the language rendition: e.g. G.711 has MOS value 4.1Greetings CCU professional Quotation:
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