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Hello,
times a matter of principle: Should an activated codec in the Sip terminal is sufficient or activate one several?
So far I always set a whole list with codecs in the terminal on active (G.711a; G.711u; G.722; G.726; ...). Is it not also sufficiently only the codec G.711a to be used? (If these from Provider, rout and terminal is supported.)
Greetings
CCU professional
That comes - as you already write down - on the Provider and your network capacity. That rout has with codecs nothing to actually do - it is, routs is also your VoIP terminal gateway.
Normally one can let G.711a and G.711u stand alone and only if necessary also different offer. Only one should also remember during mechanism of a new Providers
--gandalf.
Hello gandalf94305,
DSL line and network capacity are outer-handing available
Where I am not safe, is like the arriving and/or. outgoing discussions with VoIP Providern to be acted…
Example 1: Someone calls from its VoIP connection (e.g. 1&1) on my VoIP Festnetznummer lying (with Dus.net) on, how it holds back then if the calling (1&1) codec G.729 uses oneself and I activated only codec G.711a? Or in this case is the discussion over fixed net arrives all the same there?
Example 2: Someone calls from its VoIP connection (Dus.net) on my VoIP Festnetznummer lying (with Dus.net), how it holds back then if the calling (Dus.net) codec G.729 uses oneself and I (also Dus.net) activated only codec G.711a? In this case it would have to be a Porvider internal discussion and functions then also still?
Greetings
CCU professional
That always arrives on the Provider. Therefore there is no general rule. Some Provider can convert the discussion, others needs the same codecs on both sides. The discussion comes from the fixed net and/or. there, is present with the codecs used with ISDN anyway comparable range with G.711 goes.
In the Aushandlung a session one specifies in each case, which codecs are possible. You got problems thus only, if in the list of the offered codecs G.711 is missing. That would arrive exact on a test with dus.net, because there is no standard, which forbids or in certain situations prescribes this case.
--gandalf.
Hello gandalf,
I made evenly times a support inquiry with Dus.net on. Times sees, what me to answer…
It would be times good to know whether Dus.net servers possibly. a code conversion of the discussions know.
And here already the answer of Dus.Net:
They knnen other language codecs use around range to save or
to improve the Sprachqualitt with pure IP-Gesprchen.
In the reason the G.711a is sufficient.
To problems it should not come, there our system the language f r
Their Client to convert knows.
Dus.net can and makes thus a code conversion of the discussions!
Perhaps point also someone here in the forum, how this sees with other VoIP Providern out?
Greetings
CCU professional
Quotation:
Quotation of gandalf94305
Normally one can let G.711a and G.711u stand alone and only if necessary also different offer.
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Why do manual administer which are anyway automatically negotiated?
For so a consequence-fraught interoperability restriction SIP was not sketched.
All offer to that in descending priority sequency of the spreading/possible compatibility problems or ascending sequence of removing range depending upon requirement
own Plant.
The goal is the most favorable condition for the own plant to drive out with that the opposite side straight to still live can
Quotation:
Quotation of CCU professional
And here already the answer of Dus.Net:
To problems it should not come, there our system the language f r
Their Client to convert knows.
Dus.net can and makes thus a code conversion of the discussions!
|
Only for these codecs:
Quotation:
<-- Sip READ from 83.125.8.87: 5060: UDP
SIP/2.0 183 session progress
...
CSeq: 103 INVITE
User agent: SIP_GW_1.4.1.PL ares.dus.net
...
m=audio 51094 RTP/SAVP 8 3 2 110
a=rtpmap: 8 PCMA/8000
a=rtpmap: 3 GSM/8000
a=rtpmap: 2 G726-32/8000
a=rtpmap: 110 speex/8000
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Quotation:
Quotation of CCU professional
Should an activated codec in the Sip terminal is sufficient or activate one several?
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I recommend to keep all codecs active in the terminals, if you do not register any problems with a certain codec with _konkreten_ the terminal.
An example from own experience: Providergesprche from a mobile net arrive with my Asterisk with GSM codecs. Through internal forwarding the call goes on a Fritz box. Since also the GSM offers as codec, saves the Asterisk (that loosely the achievement for the Umkodierung would have had!) a Umkodierung and passes the Stream on as GSM.
At a terminal behind the Frotzbix had I now only one of the ISDN codecs certified, and… oh… oweh… the poor Frotzbix, whose processor is not degree the fastest, had to make now the Umkodierung.
Was only noticeable, because it always determined (GSM coded, and only detailed) telephone calls was, which were zerhackstckt. Completely apart from the fact that in the signal chain End-2-End jedwede Umkodierung should be made in such a way _selten_ as possible. Reversal conclusion: One should limit the selection of codecs only in well justified cases intentionally.
Ergo: If the terminal applies the necessary arithmetic performance, its discussions between the codecs umzumnzen correctly, should one also the terminal do let. One relieves thereby all PBXes and switching ways between them.
Quotation:
Quotation of more kritter
I recommend to keep all codecs active in the terminals.
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I had with Nokia as SIP to the FB the problem words partly checked off more rber came and longer for the cause looked.
In Nokia were 8 or 9 codecs actively, where one could naturally adapt the order and I had the codecs those the FB preferentially above.
Only after I up to 4 the other codecs rausgeschmissen has, it folds since then without problems, in my case however rather because of Nokia smells are to what…
Hello to all,
I was now last week on the Cebit and also with several Providern spoke. also these recommended to permit all normally common codecs in the terminal and VoIP TK plant and/or. to activate, so that it comes to no problems.
Greetings
CCU professional