[jitsi-dev] MTU issues with SIP + ICE


#1

I've just noticed that when I have a lot of candidate lines and a lot of
codecs in the SDP, it is easy for the INVITE packet to exceed MTU

As SIP is based on UDP datagrams, this seems most unpleasant. I've
observed OpenSIPS just truncating the packet when it is relayed (e.g.
packet received with 1326 bytes, headers added by OpenSIPS, packet goes
out at full MTU size, last candidate lines in the SDP truncated)

Has anyone else experimented with SIP and noticed such things?


#2

Hi Daniel,

Le 10/01/12 17:15, Daniel Pocock a �crit :

I've just noticed that when I have a lot of candidate lines and a lot of
codecs in the SDP, it is easy for the INVITE packet to exceed MTU

As SIP is based on UDP datagrams, this seems most unpleasant. I've
observed OpenSIPS just truncating the packet when it is relayed (e.g.
packet received with 1326 bytes, headers added by OpenSIPS, packet goes
out at full MTU size, last candidate lines in the SDP truncated)

Has anyone else experimented with SIP and noticed such things?

FYI you can use TCP or TLS for SIP signalling part to avoid this issue.

You can also reduce the payload type number of your SIP offer/answer.

Regards,

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Seb