support Asterisk 20.6 and newer versions

[finch71]

it would be nice to support Asterisk 20.6 and newer versions

[InnovateAsterisk]

The document should read 18+

It was only while Asterisk was transitioning from chan_sip to pjsip that there was a consideration on the version number. Browser Phone supports any version of Asterisk great than 18, and FreeSWITCH, or any SIP-based PBX.

[finch71]

it seems that even if i set dtlsverify=fingerprint , it still show

InvalidAccessError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.

[InnovateAsterisk]

Rather use ‘webrtc=yes’
then then don’t need to specify that kind of thing.

[finch71]

can i do this in sip.conf instead of pjsip?

[InnovateAsterisk]

It’s technically possible for chan_sip to do webrtc. I would not tho. Pjsip is a lot more mature now.

Just remember that you have to load only one sip library with load / noload in the modules.conf. They can both be loaded.

If you have to use sip.conf then you must set all the settings (can’t just use webrtc=yes)

[finch71]

thanks for the reply!

after configuring webrtc=yes, the previous error are gone but i still can not call.
now browser show no error, but on the asterisk side I get Can't provide secure audio requested in SDP offer

[InnovateAsterisk]

Capture the sip trace, and post it here. Make sure it’s the invite from the browser to asterisk.

[finch71]

 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Dec 21 05:59:59] WARNING[863][C-0000003f]: chan_sip.c:10943 process_sdp: Can't provide secure audio requested in SDP offer

the browser log shows

SessionDescriptionHandler.setDescription failed - OperationError: SIPCC Failed to parse SDP: SDP Parse Error on line 6:  Invalid port format (21418) specified for transport protocol (RTP/AVPF), parse failed.

SIPCC Failed to parse SDP: SDP Parse Error on line 6:  Invalid port format (21418) specified for transport protocol (RTP/AVPF), parse failed.

[InnovateAsterisk]

Ah, ok it’s chan_sip module struggling with it. Below is the last config I used before I stopped using chan sip.

Make sure youse is the same as mine. For the certificate stuff, make a self signed certificate.

https://github.com/InnovateAsterisk/Browser-Phone/blob/master/config/sip.conf