Step 1 Analyze logs and find that the SIP signaling is transmitted successfully. However, after the call is connected to the target IP phone, the IP phone returns a Bye message. Then, analyze the media information and find the following content, from which you can find the IP phone does not support the Secure Real-Time Transport Protocol (SRTP) streams.
[0x0000070e][2013-02-05 10:45:09.94][conference.c 19210] CONF_ReleaseSeat:::[dwSeatNo = 65], Seat not suppot srtp!!!
[0x0000070e][2013-02-05 10:45:09.94][conference.c 6852] CONF_ReleaseSeat:::[dwSeatNo = 65]
[0x0000070e][2013-02-05 10:45:09.94][conference.c 5870] CONF_SendQ931MsgToCCM:::[dwSeatNo = 65][dwCCMNo = 11][MsgType = EN_Q931_MSG_RELEASECOMPLETE]
[0x0000070e][2013-02-05 10:45:09.94][conference.c 19379] CONF_ReleaseConfResource:::[dwSeatNo = 65]
Step 2 Check the voice encryption setting for the IP phone. By default, the value is Off, indicating that the Real-Time Transport Protocol (RTP) is used, as shown in Figure 3-1.
Figure 3-1 Voice encryption disabled
Step 3 Check the encryption setting of the scheduled conference and find that the option Encrypt is selected for Confer SRTP by default, as shown in Figure 3-2.
Figure 3-2 Encryption mode enabled for the scheduled conference
Step 4 Set Voice Encryption (SRTP) to On for the IP phone. Then, make a call to the IP phone using the eSpace U1910 self-service system. The IP phone user successfully joins the conference. The fault is rectified.
----End