Don't expect a TCP transport to be working for media on high latency networks such as EDGE or GPRS, or even on networks with packet loss (just 0.5% BER will kill the flow) where TCP congestion will occur and block the audio flow. Fabio On 26/07/10 18.01, Shrouk Khan wrote: > Hi guys, > I am working on pjsip on symbian . I am trying to pack all the SIP and > RTP data generated by pjsip into a tcp packet and send it to a single > port proxy server . >
