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Knowledge Base

Call Initiated from the eSpace U1910 Web Self-Service System Is Automatically Ended Upon Off-hook

Publication Date:  2015-07-24  |   Views:  937  |   Downloads:  0  |   Document ID:  EKB1000073608


Issue Description

A conference has been scheduled using the eSpace U1910 web self-service system. When the scheduled time arrives, the eSpace U1910 web self-service system makes a call to an IP phone. The IP phone rings. When the IP phone is picked up, the call is ended immediately.

Handling Process

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.


Root Cause

No voice encryption is configured for the IP phone.


The terminal used to a join an encrypted conference must support voice encryption and has the function enabled.

The integrated access device (IAD) does not support SRTP encryption. In the scenario where the eSpace U1900 series unified gateway is deployed with IP phones and the IAD is deployed with analog phones, you must disable SRTP encryption. If SRTP encryption is enabled, analog phones under the IAD cannot join the conference because the IAD does not support SRTP encryption.

Common support for encrypted IP phone includes: eSpace 7800 series, 7900 series and 8950.