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Codec interworking from analog lines to voice

Dear community,

I am new to ip telephony and voice over ip in general. I am currently working in a setup connecting two routers between each other and attaching one of them to the pstn over two FXO connection and the second router has two analog phones. I need to know how the codecs work in this setup since the analog phones use G.711 by defaul and the dial-peer between the routers use g728 by default. I am wondering how in this case the communication is possible. Is there some transconding behind the scene that I am not aware of? I am not usind codec classes and I am not forcing the dial-peer to use g711 but it is working. The routers I am using are 2811. I will really appreciate if you can point me some information to help me understand how codec works from analog to ip and viceversa.

Best Regards,

Rafael.

1 ACCEPTED SOLUTION

Accepted Solutions

Rafael,

Nothing is using G711, you have only 2 call legs, POTS leg which is TDM signaling in this case analog line, which uses TDM PCM encoding (you can compare it to G711) and one voip call leg which is using G729. The transcoding is taking place on the voice port itself when TDM signalling is converted to voip signalling, but keep in mind there is no voip transcoding from one codec to another in this scenario.

DSP farms would be required if you needed to transcode voip call legs, for example when 2 endpoints cannot negotiate codec, a good example is an IP-IVR system that can be defined with only one codec, you would need to transcode that if calls are set to use another codec.

HTH, please ratea all useful posts!

Chris

View solution in original post

3 REPLIES 3
Chris Deren
Hall of Fame Master

The voice ports (FXO) and (FXS) can negotiate either codec so as the call is being setup the voip protocol in use (i.e. H323, SIP) exchanges the port's capabilities and they agree on a codec. As you said the default dial-peer codec is G729 and unless you explicitly define other codec, i.e. G711, then it will be used.  Keep in mind that you should be using g711 unless network bandwidth is a concern here, G729 will compress the call and use less bandwidth, but it will use two PVDM (DSP) channels as oppose to 1, so you need to ensure you have sufficient number of PVDMs on both routers.

HTH,

Chris

Hello Chris,

Thanks for answering in so short time. So just to understand clearly, in my case the call uses g729 because one of the call legs is using g729 (the dialpeer voip in this case). The endpoints which are using g711 by default end up using g728, but some transcoding is still required in this case. I read somewhere it was necesary to configure DSP farms for transcoding in the routers, and you told me the DSP transcoding is done automaticaly, so I guess the new routers are able to accomplish this transcoding without manual configuration, is that right?

Best Regards,

Rafael

Rafael,

Nothing is using G711, you have only 2 call legs, POTS leg which is TDM signaling in this case analog line, which uses TDM PCM encoding (you can compare it to G711) and one voip call leg which is using G729. The transcoding is taking place on the voice port itself when TDM signalling is converted to voip signalling, but keep in mind there is no voip transcoding from one codec to another in this scenario.

DSP farms would be required if you needed to transcode voip call legs, for example when 2 endpoints cannot negotiate codec, a good example is an IP-IVR system that can be defined with only one codec, you would need to transcode that if calls are set to use another codec.

HTH, please ratea all useful posts!

Chris

View solution in original post

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