We encounterred with SIP signaling commands lost issues recently in different terminals, environments, and scenarios.
And we were using UDP as our prior transport type.
The potential cause could be:
1. There were SIP commands which could larger than MTU size.
2. The send/recv queue buffer size of the socket handle was not enough.
3. SIP command(Conference control) were really tremendous

There are some informations about this issue, also could be a way out of this.

According to ​RFC 3261 section 18.1.1:
“If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 congestion controlled transport protocol, such as TCP.”

if Request is Larger than 1300 bytes.

By this rule, PJSIP will automatically send the request with TCP if the request is larger than 1300 bytes. This feature was first implemented in ticket #831. The switching is done on request by request basis, i.e. if an initial INVITE is originally meant to use UDP but end up being sent with TCP because of this rule, then only that initial INVITE is sent with TCP; subsequent requests will use UDP, unless of course if it’s larger than 1300 bytes. In particular, the Contact header stays the same. Only the Via header is changed to TCP.
It could be the case that the initial INVITE is sent with UDP, and once the request is challenged with 401 or 407, the size grows larger than 1300 bytes due to the addition of Authorization or Proxy-Authorization header. In this case, the request retry will be sent with TCP.
In case TCP transport is not instantiated, you will see error similar to this:
“Temporary failure in sending Request msg INVITE/cseq=15228 (tdta02EB0530), will try next server. Err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
As the error says, the error is not permanent, as PJSIP will send the request anyway with UDP.
This TCP switching feature can be disabled as follows:
● at run-time by setting pjsip_cfg()->endpt.disable_tcp_switch to PJ_TRUE.
● at-compile time by setting PJSIP_DONT_SWITCH_TO_TCP to non-zero
You can also tweak the 1300 threshold by setting PJSIP_UDP_SIZE_THRESHOLD to the appropriate value.