no sound one way with SIP snom m3 phone

[fleurys]

I have an snom m3 phone with the latest firmware 2.11

My router in a juniper ns5gt. I have read the basic guide about the m3 on this website.

I can get my phone connect no problem with fpl. I can call from it and land line can also call to it.. Up to there no problem.

Where the problem is, is that on both cases, the landline can hear my voice, but I do not hear theirs... So communication is always one way : M3 to landline = ok landline to m3 = silence.

Now I know it has to do with NAT or port forwarding. I have been playing with this for the last 5 hours and have not been able to advance. My setup is simple. local lan, router, internet.

I see in the m3 server log that the rtp port is establishing itself at 5004 udp... now on the faqs on fpl they talk about udp port 13000,13001,5060,5061,6060,and 6061. I have tried to forward all of these plus the 5004 to the sip server to no avail... But i,m not even seeing traffic going to it, so most probably still an issue with router... Anyone here has any experience with juniper ns5gt and sip phone with fpl ? I have read multiple webpages with people struggling with this router and voip, but i'm trying to see if someone here has seen this kind of behavior before I spend another xx hours trying stuff...

Tks.

[fleurys]

Well, looks like luck went my way.... I found this gem right here : https://teleconnex.zendesk.com/entries/ ... figuration


One little thing at the beginning that says : Server is local - YES (if this is set to No, we experience one way audio issues.)

So for me, with the router I have, I only needed to set SERVER IS LOCAL to YES and that was it. I did not have to even have any port forwarded. nothing... no firewall policies etc.... basically a plug and play for this router and set the m3 with this setup eventhough in the fpl guide for the m3 is states to put at NO... so I guess this setting could be either depending on the router you have.... no wonder I was not getting anywhere with all my 38839928392 tries !

S.

[bridonca]

When you have one way audio issues, you do not have a reliable route to RTP from the internet to the ATA. SIP VOIP, to work, needs a SIP port, (usually port 5060, but with freephoneline, you can also select UDP port 13000,13001, and ports 5060 to 6060 as SIP ports), and at least one RTP port, which can be any available UDP port. I usually forward ports 12000 to 12007 from the router to the ATA for RTP. I am not sure what you did to make it work, but I suspect it will not be reliable.

[fleurys]

I dont feel the same way as you do... looking at the logs of the router, it clearly sees the packets coming as sip protocol and does not change the port passing throug nat... This router is not something you buy at the corner store.... it's a fullfledge enterprise device.... The connection is extra stable and the quality is perfect.... so ... sometimes, with the right stuff, it CAN be easy.... I