Re: [SR-Users] How to disable retransmits via TCP connection? etc.

2015-06-19 12:45 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>: Thank you Daniel for answering. We still struggle from this weird issue. Now I can reproduce it in my location with recent Kamailio from git, with all SIP over TCP clients - Linphone, Sipdroid, Jitsi (desktop Java-based app). Here are ngrep logs from both server and client sides, and server syslog containing Kamailio output (filtered by substring "tcp" case-insensitively). https://gist.github.com/krieger-od/55427f2b3923b910bacb https://gist.github.com/krieger-od/c9fe6ea4bb64fac82cda https://gist.github.com/krieger-od/96ef40ca15ef2407b5f4 Here you can see that only a "tail" of INVITE message gets transmitted to client side, which is weird. This is Amazon server; it have MTU 9001 by default, but setting it to 900 or 1100 haven't made any difference. Also I have reproduced this issue on DigitalOcean VPS, so this mustn't be Amazon-specific issue. Because I have tried different SIP useragents supporting TCP, I'm afraid this can be considered Kamailio issue (honestly, I still don't quite believe as I percept Kamailio as robust and stable software). Any review and comment helps. -- Andrey Utkin