[BUG]Call-ID garbled when start with 's'

[CzyerChen]

OpenSIPS version you are running

version: opensips 3.4.9 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: cd02af8d9
main.c compiled on 15:28:56 Nov  6 2024 with gcc 9

Describe the bug

When using Sipp to test or produce freeswitch simulation, and the Call-ID field is set to the beginning of s, for example, s1001, opensips starts and debug mode is enabled, and you will see that the printed signaling content $ci is garbled data.

To Reproduce

1. Start OpenSIPS
2. Start traffic with special headers
3. Inspect traces
4. Notice the Call-ID has wrong format in logs, and opensips failed to process messages within dialog.

sipp:

sipp ip:5060 -sf uac.xml -d 10000 -s phonenum  -r 1 -rp 1000 -m 1 -cid_str as%u-%p@%s  --- Call-ID like [email protected], testcase passed.

sipp ip:5060 -sf uac.xml -d 10000 -s phonenum  -r 1 -rp 1000 -m 1 -cid_str a%u-%p@%s  --- Call-ID like [email protected], testcase passed.

sipp ip:5060 -sf uac.xml -d 10000 -s phonenum  -r 1 -rp 1000 -m 1 -cid_str %u-%p@%ss  --- Call-ID like [email protected], testcase passed.

sipp ip:5060 -sf uac.xml -d 10000 -s phonenum  -r 1 -rp 1000 -m 1 -cid_str s%u-%p@%s  --- Call-ID like [email protected], testcase failed.

logs in file:
correct in INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.1:5061
From: "sipp" <sip:[email protected]:5061>;tag=1
To: <sip:[email protected]:5060>
Call-ID: [email protected]
Cseq: 1 INVITE
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Content-Length:   139

v=0
o=user1 53655765 2353687637 IN IP4 192.168.5.1
s=-
t=0 0
c=IN IP4 192.168.5.188
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000

Garbled in reply:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKb37.1c94a6b.0
From: <sip:[email protected]>;tag=1
To: <sip:[email protected]>;tag=4
Call-ID: ?H???????
CSeq: 1 INVITE
Contact: <sip:192.168.5.1:5081;transport=UDP>
Content-Type: application/sdp
Content-Length:   195

sipp uac.xml:

<?xml version="1.0" encoding="UTF-8" ?>
<scenario name="Basic UAC scenario">
 <send>
  <![CDATA[  
  INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port]
  From: "sipp" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[service]@[remote_ip]:[remote_port]>
  Call-ID: [call_id]
  Cseq: 1 INVITE
  Contact: <sip:sipp@[local_ip]:[local_port]>
  Content-Type: application/sdp
  Content-Length: [len]
  
  v=0
  o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  s=-
  t=0 0
  c=IN IP[media_ip_type] [media_ip]
  m=audio [media_port] RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  ]]>
 </send>

 <recv response="100" optional="true">
 </recv>

 <recv response="180" optional="true">
 </recv>


 <recv response="200">
 </recv>

 <send>
  <![CDATA[
  ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port]
  From: "sipp" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  Cseq: 1 ACK
  Content-Length: 0
  ]]>
 </send>

<pause milliseconds="10000"/>

 <send retrans="500">
  <![CDATA[
  BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port]
  From: "sipp" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  Cseq: 2 BYE
  Content-Length: 0
  ]]>
 </send>

Expected behavior

Not garbled in reply.

Relevant System Logs

DBG:dialog:get_dlg_by_callid: input ci=(14)
rm=[INVITE],ci=[?H???????],rs=[180]
rm=[INVITE],ci=[?H???????],rs=[200]

OS/environment information

• Operating System: CentOS 7.9
• OpenSIPS installation: 3.4.9
• other relevant information:

Additional context