Since your subject says without SIP signalling update, no. The phone needs to detect its IP change and tell FS. FS needs to know where to send signalling messages to as well as media - ringing, answer, hangup etc. That means it needs to know the new IP & port for the SIP part too. The closest to what you're asking is RTP auto-adjust, which is a workaround for devices unable to handle NAT correctly. But I don't think it should allow it mid-call, or that could allow someone to hijack your phone call. -Steve On 21 March 2013 20:44, Anthony Minessale <anthony.minessale@gmail.com>wrote:
yes if there is a re-invite it will.
On Thu, Mar 21, 2013 at 2:40 PM, Jyotshna Cherukuri < jcherukuri_necc@yahoo.com> wrote:
Hi,
I am wondering if current version of FS has the capability to dynamically switch RTP trasnmission to the source address of RTP packets if the source address of RTP packets gets changed automatically while a call is already in progress . This might happen when a mobile handset gets a different IP on a access point reconnect during an ongoing call.
Any help is greatly appreciated
Thanks in advance Regards Jyotshna
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