Rtp audio sent to wrong pjsip endpoint Contact ip

I suspect i’m having an issue with the pjsip ps_endpoints options rewrite_contact and rtp_symmetric. Long story short, sip phone behind ipv4/ipv6 translation. SIP Contact headers contain the ipv6 address but the real source from asterisk point of view is the ipv4 address. The asterisk “pjsip show endpoint” shows proper ipv4 in the “Contact:”, and the SIP call setup works, ie. phone rings, and you can dial out. But once the audio starts, on the asterisk cli you see an error for trying to connect to the ipv6 address which is not possible as the asterisk is on ipv4.

WARNING[4801] acl.c: Cannot connect to 2605:b100:349:5cd1:0:4:375a:3e01: Cannot assign requested address

Contact headers aren’t use for media. The c= line in the SDP is used for that.

thanks. the c= line does contain the ipv6.

I’m using GSwave on samsung. i switched back to my old samsung and GSwave uses the ip4v ip instead of the ipv6 and it works. Switched back again to new GSwave/samsung and back to ipv6 ip and no audio. ie. no RTP. Issue must have something to do with GSWave and ipv6? I don’t see any options to force use of ipv4.

Welcome! Agreed! And/or issue with your Asterisk instance not listening on IPv6.

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