By studying WebRTC sources I was wondering why developers chose max RTP packet size as 1200 bytes. It's defined in media/engine/constants.cc, as kVideoMtu variable.
Thanks.
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To avoid IP fragmentation. – aergistal Dec 04 '17 at 20:27
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Thanks, but why they didn't choose 1150 or 1250. I.e. what magic lies under the hood of this constant? – Taras Koroliuk Dec 05 '17 at 08:56
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Minimum MTU for IPv4 is 68 bytes but much larger in reality and 1280 bytes for IPv6. Typical link MTU is 1500 but packets can be tunneled. The 1200 bytes value works for 99% of the cases with both IPv4 and IPv6. – aergistal Dec 05 '17 at 10:04
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2Maybe someone find this link useful https://groups.google.com/forum/#!topic/discuss-webrtc/gH5ysR3SoZI – Taras Koroliuk Dec 05 '17 at 14:49
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This is an arbitrarily selected value to avoid packet fragmentation. There is no any exact science behind this as you can be never sure on the actual limits, however 1200 byte is a safe value for all kind of networks on the public internet (including something like a double VPN connection over PPPoE) and for RTP there is no much reason to choose a bigger value.
Istvan
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1For IPv6 you can be pretty damn sure, it's 1280 in the RFC and one of the reasons for the chosen value. – aergistal Dec 05 '17 at 10:06
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Here is a good explanation why packet should be lesser or equal than 1280 bytes for IPv6 networks and why we need to avoid packet fragmentation in IPv6 networks https://www.arbornetworks.com/blog/asert/ipv6-fragmentation/ – Taras Koroliuk Dec 07 '17 at 11:30