I'm working with freeswitch and I made the connection between my server and another one, for hearing each other I used the codec G729. The issue is the next: I call them, the call is established and I can hear the other part perfectly but they can't hear me and they can see the audio packets coming to their server but they can't hear me, any idea? Thanks!
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2 Answers
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I'd start by looking at the firewall on the remote machine. libpcap will allow you to see all packets hitting the box before the firewall rules are applied, so it can be a little deceptive. So there's a good chance that the firewall is blocking the packets.
Also, you didn't specify what the other user is connecting to their server with, and how. So if they're connecting to their server from behind NAT, then you could just be experiencing the typical one-way audio issues so common with NAT scenarios. Check out http://wiki.freeswitch.org/wiki/Nat for more info.
This issue doesn't seem to be directly related to the g.729 codec or you'd have most likely gotten a 488 SIP response.
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It's possible that the person on the other end isn't able to decode the the G729 stream.
What kind of phone are they using?
I would google that phone model and try a known supported codec for it. Then I would setup your Freeswitch to encode for that codec.
If that works then it's likely an issue with the phone.
Hope this helps.
