[Freeswitch-dev] Freeswitch's capability of switching RTP transmission to new destination's address without any SIP signaling update
steveayre at gmail.com
Fri Mar 22 00:07:37 MSK 2013
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.
On 21 March 2013 20:44, Anthony Minessale <anthony.minessale at 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 at yahoo.com> wrote:
>> 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
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> Official FreeSWITCH Sites
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
> Anthony Minessale II
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