if ($var(isTpgw) == 1)
{
xlog("L_INFO","[$ci][$mi][CSeq $cs] isTpgw = $var(isTpgw)");
if (!t_relay_to_tcp())
{
xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp....Error...\n");
t_on_failure("DISPATCHER_FAIL");
}
else
{
xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp\n");
}
}
But when I looked at my sip capture tool, it showed the Via marked by the Kam to use UDP:
INVITE sip:######@<some domain>:5060 SIP/2.0 Record-Route: <sip:206.81.191.30;r2=on;lr=on> Record-Route: <sip:206.81.191.30;transport=tcp;r2=on;lr=on> Call-ID: DL9dc0c8dd93-1079779342@gs-3924.GSLAB.COM To: "sip:6204857@sjomaintpsg50.fuzemeeting.com" <sip:6204857@sjomaintpsg50.fuzemeeting.com> From: "LifesizeSIP" <sip:LifesizeSIP@gs-3924.GSLAB.COM:5060>;tag=DL325a9e10c7;epid=062EE468 CSeq: 1 INVITE Max-Forwards: 69 Via: SIP/2.0/UDP 206.81.191.30;branch=z9hG4bKb16b.bd6a0791c798e3c859f8608993d584ae.1;i=5611 Via: SIP/2.0/TCP 10.101.199.105:5060;rport=53272;branch=z9hG4bK-fc8090ecf8-DL
Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342@gs-3924.GSLAB.COM][42][CSeq 1] isTpgw = 1
Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342@gs-3924.GSLAB.COM][42][CSeq 1] LifeSize Softphone 8.1.12 (Windows) Relaying [INVITE] from (10.101.19
9.105) to (206.81.191.72) via tcp
So why did the Kamailio mark the Via header on the outgoing INVITE UDP?