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I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details:

encryption = yes
avpf = yes
icesupport = yes
dtlsenable =yes
dtlsverify = no
dtlssetup = actpass
dtlscertfile = valid path to certyficate
dtlsprivatekey = valid path to certyficate
force_avp = yes

And all is working with webphone. But when I want to call from my peer (with config above) from softphone (like zoiper or linphone) I received an error "488 Not acceptable here bearer capability not presently available" and in Asterisk console:

[Oct 27 12:00:23] NOTICE[2214][C-00000001]: chan_sip.c:10148 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 8000 RTP/AVP 3 110 8 0 98 101
[Oct 27 12:00:23] WARNING[2214][C-00000001]: chan_sip.c:10543 process_sdp: We are requesting SRTP for audio, but they responded without it!

And my question is simple.. Can I configure Asterisk peer that it can work with softphone and webphone? :)

Thanks in advance! Matt.

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This isn't a problem with connecting softphones with webphones. The user agent (I assume the one you wrote - but you don't specify) is not responding properly. The other UA (or soft switch inbetween) is trying to establish an RTP connection. One UA said it supports SRTP (secure) but it didn't use a secure RTP channel. That is confusing the other UA. And so I suspect the call breaks down (though you don't show any details of that).

Solution is likely to fix the UA to either not broadcast that it supports SRTP, or have it implement SRTP.

Based on the 2 lines you posted above you are using Asterisk as the soft switch. I would suggest you enabled RTP and SIP debug on Asterisk, and ensure the UA's don't allow reinvites (also called Direct Media). That will force the RTP stream through Asterisk and you will see the headers clearly showing the mismatch.

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