Is it possible to establish a SIP session without initial Audio

I have a SIP device that initates a call with only a video stream offered in SDP. When a talk button is pressed, the device will send a re.INVITE with audio and video in SDP. This works fine with other SIP Server we used. but it fails with Asterisk.

So my question is it possible to convince Asterisk (maybe with settings in sip.conf) to accept calls without audio stream? We currently use chan_sip.

When the first INVITE comes in (without audio in SDP) Asterisk complains with following error:

[Dec 14 13:23:35] ERROR[29492][C-000000ef]: translate.c:1393 ast_translator_best_choice: Cannot determine best translation path since one capability supports no formats
[Dec 14 13:23:35] WARNING[29492][C-000000ef]: channel.c:6179 request_channel: No translator path exists for channel type SIP (native (codec2|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|silk|silk|silk|silk)) to (h264)
[Dec 14 13:23:35] WARNING[29492][C-000000ef]: app_dial.c:2507 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 58 - Bearer capability not available)

I don’t believe that chan_sip allows media streams to be added at all, although a stream can be started in an inactive state.

I don’t know about PJSIP, although Asterisk is a voice PABX with video bolted on, so it is quite possible it won’t like not having an audio stream, either.

PJSIP would have a better chance in the future of supporting such a thing, but the Asterisk core itself really wants an audio stream to be present.

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Thank you for the information. I changed the UAC implementation to start immediately with audio. That works.

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