[Asterisk-Users] Call between G.711 and GSM

Jamie Carl me at jazz-inc.net
Mon May 19 16:25:05 MST 2003


Hmm...  I've only ever seen it dump an error saying "No 
Compatible Codecs" in these circumstances.

Recompression would be good for compatability.  But poor 
for performance.  Where has the line been drawn?

Jamie Carl
me at jazz-inc.net

On Mon, 19 May 2003 14:22:39 +0300
  "Dan" <dtoma at fx.ro> wrote:
>*This message was transferred with a trial version of 
>CommuniGate(tm) Pro*
>Hi,
>
>> *This message was transferred with a trial version of 
>>CommuniGate(tm) Pro*
>> Will asterisk actually convert between two different 
>>codecs?????
>> ie, a SIP endpoint running GSM and another running 
>>G.711?
>It seems yes, if the endpoints does not have a common set 
>of codecs
>available.
>
>>
>> Wouldn't that add quite some latency?
>Yes.
>
>> I was always under the impression
>> Asterisk did not recompress and was smart enough to 
>>negotiate the right
>> codec at each end and just pass through the RTP packets.
>It is, but when the two devices uses different codecs 
>then.. what to
>negociate?
>When you register the two SIP phones on another SIP proxy 
>(like FWD) then
>when you try to call each other the call cannot be 
>established. In Asterisk
>it can.
>
>BR,
>Dan
>P.S. I'm very new in the Asterisk world, so please be 
>patient with me...;-)
>
>>
>> Regards,
>>
>> Jamie Carl
>> Email: me at jazz-inc.net
>> PH: +61-414-365-466
>>
>> -----Original Message-----
>> From: Tjardick van der Kraan 
>>[mailto:tjardick at vanderkraan.net]
>> Sent: Monday, 19 May 2003 9:00 PM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [Asterisk-Users] Call between G.711 and GSM
>>
>>
>> *This message was transferred with a trial version of 
>>CommuniGate(tm)
>> Pro*
>> The GSM codec in X-Lite is not compatible with the GSM 
>>codec used in *.
>>
>> I know X-ten is working on a new versio with a different 
>>code instead of
>> GSM...
>>
>> Greetings,
>>
>> Tjardick
>>
>> ----- Original Message -----
>> From: "Dan" <dtoma at fx.ro>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Monday, May 19, 2003 12:36 PM
>> Subject: [Asterisk-Users] Call between G.711 and GSM
>>
>>
>> > Hi all,
>> >
>> > I have a Cisco 7960 IP PHONE and an X-Lite soft phone, 
>>both connected
>> to
>> an
>> > Asterisk PBX as SIP phones.
>> > If G.711 codec is used by both phones (GSM codec 
>>disabled on X-Lite),
>> the
>> > sound is perfect.
>> > When I enable GSM on X-Lite, it tends to use it for 
>>any call and then
>> > Asterisk make a conversion between GSM and G.711 (I'm 
>>right with this
>> > one???).
>> > The call is established, 7960 display the used codec 
>>as G.711 and
>> X-Lite
>> as
>> > GSM, but all I can hear is only a lot of noise. The 
>>music on hold
>> feature
>> > works ok on both sides.
>> >
>> > How can I do to use this type of configuration?
>> > 7960 does not have GSM codec, nor G.723 and I have no 
>>G.729 on X-Lite.
>> > I want to be able to use it where G.711 is not 
>>suitable because of the
>> > available bandwidth.
>> > Cisco 7960 is in the same LAN as Asterisk PBX, but 
>>X-Lite client ie
>> > remotely, over a not so good link (but perfect for GSM 
>>codec).
>> > Any suggestions?
>> >
>> > Thanks,
>> > Dan
>> >
>> >
>> > _______________________________________________
>> > Asterisk-Users mailing list
>> > Asterisk-Users at lists.digium.com
>> > 
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>>
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