Hi I am facing audio issue one way. I am running PJSIP one side. I am using Kamailio 3.1.5 and RTP proxy as an intermediate proxy (both for signalling and media) and I have a main proxy (having registration , accounting, routing support). When I make a call from PJSIP to a mobile phone , maximum times I notice audio quality in PJSIP to Mobile phone direction is very bad. However audio from Mobile phone side to PJ phone is side is very good. On further debugging, I found that the intermediate Kamailio proxy modifies SIP messages and add NAT specific attributes in SIP/SDP. Did anybody face this issue anytime? If you have faced , please share how could we overcome this issue. Thanks Protea -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20111010/0cdab621/attachment.html>
