Changeset 4177 for pjproject/trunk/tests/pjsua/scripts-sipp/uas-invite.xml
- Timestamp:
- Jun 26, 2012 2:28:59 AM (12 years ago)
- File:
-
- 1 edited
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pjproject/trunk/tests/pjsua/scripts-sipp/uas-invite.xml
r2575 r4177 47 47 [last_Call-ID:] 48 48 [last_CSeq:] 49 Contact: <sip: 192.168.0.15>49 Contact: <sip:sipp@[local_ip]:[local_port]> 50 50 Content-Type: application/sdp 51 51 … … 67 67 </recv> 68 68 69 <!-- Keep the call open for a while in case the 200 is lost to be -->70 <!-- able to retransmit it if we receive the BYE again. -->71 <pause milliseconds="4000"/>72 73 69 74 70 <!-- definition of the response time repartition table (unit is ms) -->
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