Common SIP, RTP and SDP Implementation Errors

RTP-Related Bugs

Support of any packetization time. Some implementations restrict the max input audio packet size to 20 ms or 30 ms.
If a test application just loops back the audio packet, then it should be careful not to loop back the RTP headers also (especially the SSRC). Loop back should be done at the media level.

One implementation responded to requests with Via headers that had been reversed, causing the response to go right to the sender.
One implementation ignored route headers when the request URI had an maddr parameter.
Use the Expires in "Contact:" if present instead of "Expires:". Aso, when retrying register refresh, an UA should handle error cases (i.e., timeout <=0 ).
Case insensitivity of stale param in WWW-Auth. stale=FALSE is not interpreted correctly.
Proxy forwards request to other address if the domain in requestURI does not belong to itself.
Distinction between 404 and 480?
Send CANCEL or BYE if 200/ACK was not received or sent properly (or timed out).
Send ACK every time you receive (even retransmission of) final response of INVITE.
Set Content-Length in TCP SIP request and response; if it is not set the things do not work as expected.
Some implementation did not send back the response to the port number specified in Via, instead used 5060 always.
Blank spaces, tabs, multi-lines for WWW-Authenticate header.
Retransmitted responses different from the original response - they must be same.
Add appropriate tag in Via if the IP address does not match the claimed Via.
Sending an OPTIONS method in the middle of a call causes the call to fall over.
Confusion between the two CSeq numbering spaces.
Case sensitivity with header and token names.

For gateway implementation (which can support only one audio codec at a time), the gateway sends list of codecs in INVITE but expects the remote UA to reply with only one (selected) codec in m= line. Correct way is to use re-INVITE.
Ordering of t= line.
Some implementation did not look into the payload type of RTP packet. Assumed to be 0 (PCMU/8000).

Last updated by Henning Schulzrinne