SIP And Voice On Different Nets
I have connection with two networks (by VoIP provider setup)
1 – 10.10.10.0/24 = SIP
2 – 10.10.11.0/24 = Voice
How to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voice
Unfortunately, I _need_ to use two networks instead of one
4 thoughts on - SIP And Voice On Different Nets
Both the chan_sip and chan_pjsip modules have a “media_address” option which can be used to specify the address to place in the SDP for media. In the case of chan_pjsip there is also a “bind_rtp_to_media_address”
option which can be used to guarantee that RTP leaves from that same address as well.
Cheers,
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Joshua Colp Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW – Huntsville, AL 35806 – US
Check us out at: http://www.digium.com & http://www.asterisk.org
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By voice do you mean RTP? Are you using chan_sip or pjsip?
Yes, Voice = RTP
Using chan_sip
2017-04-27 15:32 GMT+03:00 Dovid Bender:
Seems I responded the same time as Josh. Follow what he has suggested.