I am having a similar problem with SIP UA based on PJSIP. My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work. What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ... When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ... For all calls i am using inbound-proxy-media=true . Is the same issue you see/encounter? Fabio On 23/03/10 17.26, Madovsky wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks