Pete, As you saw the SIP MTU issue is a complex one; I get into that in my blog post here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html Needless to say I was "inspired" by Apple's use of HTTP-like SIP body compression and it's now been implemented in Kamailio and FreeSWITCH: http://kamailio.org/docs/modules/4.1.x/modules/gzcompress.html http://jira.freeswitch.org/browse/FS-5814 So if you're using endpoints that support it you may be able to include many more codecs without running into the IP fragmentation issue. Also (as I note in my post) TCP suffers from far fewer issues with fragmented IP packets in practice so you could just do that (or both). Regarding codecs themselves, it's important to note that while FreeSWITCH and other OSS projects support a wide range of codecs often times commercial equipment does not. Unfortunately it's this commercial equipment that's often at the actual ends of a call (IP handset to LEC PSTN gateway, for example). The large scale media gateways manufactured by Cisco, Sonus, and others in use by the various LECs that operate the actual PSTN gateways often times only support G711a, G711u, and G729. The use of any other codec requires transcoding when using these gateways and transcoding is never a good thing - it increases delay and reduces quality. Commercial IP handsets aren't much better. Your options for 8k codecs are about the same as above and for wideband (16k or better) you'll see support for G722 and others. WebRTC provided a huge shot in the arm for OPUS, which can actually run from 8k - 48k. Unfortunately it's still not supported by any existing media gateways or handsets that I'm aware of. On Sat, Mar 22, 2014 at 9:40 AM, Pete Ashdown <pashdown@xmission.com> wrote:
So considering I can't put every possible codec under the sun into codec_prefs, due to the aforementioned MTU limit on the SIP packet, what is considered an optimum set of codecs that fits with a wide compatibility base?
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