Call Drop due to RTP Timeout due to port misconfig

Call drops are natural especially in countries like India where congestion is so evident under the radio environment. There are many reasons for a call drop but out of all RTP Timeout is stands top among all.

RTP – Real time protocol plays important role in delivering speech packets during a call. all these use UDP as transport as speech packets are time critical rather than reliability.

Below scenario is one which can lead to RTP timeout. Here while call setup is in progress and SIP invite triggered at IMS which will make the baseband to establish the dedicated bearer. as part of it Network sent Activate dedicated EPS bearer context request with all the required configuration embedded.

For example as shown below the traffic flow template from network in uplink direction local port is 3400 and in downlink remote is 3400 which becomes 3400 port for UE in both uplink and downlink directions.

Instead in uplink if local port is 3400 and in downlink remote port should be 3456 not 3400 again. Because of this incorrect config MT device will be waiting for the RTP flow where as MO device will not able to send the packets to the target leading to RTP timeout at the MT device and call drops inevitably.

Below is the SIP Invite where under media line device attached IP address and port details received from Activate dedicated EPS beared request from network but since the config is incorrect it is sending with port 3456 instead of 3400 which is local port for UE in uplink direction.

INVITE tel:+1-212-555-2222 SIP/2.0
Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 70
Route: <sip:pcscf1.home1.net:7531;lr;comp=sigcomp>, <sip:scscf1.home1.net;lr>
P-Preferred-Identity: “John Doe” <tel:+1-212-555-1111>
P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id-3gpp=234151D0FCE11
Privacy: none
From: <sip:[email protected]>;tag=171828
To: <tel:+1-212-555-2222>
…..

…..

…..
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE
Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi-c=98765432; spi-s=87654321;
port-c=8642; port-s=7531
Content-Type: application/sdp
Content-Length: (…)

v=0
o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
s=-
c=IN IP6 5555::aaa:bbb:ccc:ddd >>>>>>>>>>> Connection type – it shows the I-address to be used for this transaction
t=0 0
m=audio 3456 RTP/AVP 97 96 >>>>>>>>>>>> Media line – shows local port used by the UE to PCSCF followed by the codec for the speech packets(RTP)

Source : 24.228 and 24.301