There is no such thing as trunking in sip as in trunking like iax2... There are much easier ways to deal with this in sip... For instance set your ptime between freeswitch boxes to be 40ms, this effectively cuts the RTP/PPS overhead in 1/2... Set it to 60ms cut it by 66%.... This works by increasing the amount of audio sent in 1 pack from 20ms to 40 or 60ms... Merging RTP between calls after that point is pretty much a mute point as you already are achieving a large reduction in RTP overhead... K On 1/28/12 2:42 PM, "John" <freeswitch@earthspike.net> wrote:
It does make a considerable difference if you are running your RTP through IP encryption (eg IPSec VPN) then the per-packet overhead becomes a significant factor. Companies such as NET (www.net.com) provide switches (VX400, VX900, etc) that run bespoke SIP-like and RTP-like protocols that merge RTP calls between 2 WAN nodes into single packets and this can reduce the per-packet overhead of encryption quite significantly. The cost is a small additional latency (eg 20-40ms) while the packets from various calls are aggregated.
John
PS. I have no affiliation with NET.
On 28/01/12 20:26, georg@riseup.net wrote:
SIP only has gateways, no trunks. When you're dealing with VoIP, treat SIP like data, not like voice channels. Allright, but makes this a difference in practice?
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