xij*_*dai 1 sip freeswitch sdp
我是sip/sdp世界的新手.
根据我对SDP协议的理解,如果我们将sip服务器中的a = sendonly定义为客户端软电话,则软电话应打开一个RTP会话进行监听,但不应将任何RTP数据包发送到目的地.我对么?
在我的情况下,我听不到任何声音,并且有一个RTP流来上传音频.注意:我正在使用多播地址.
这是SIP/SDP转储(从服务器到客户端软电话):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.219:5060;branch=z9hG4bK-d8754z-b394381274917501-1---d8754z-;rport=5060
From: ;tag=d67855ee
To: ;tag=KQQHgQ93Sjg1F
Call-ID: YTExMzkwZDdhMGM1NTJmMDJlMGFiYjgxMGI1ZDNmMDI.
CSeq: 2 INVITE
Contact:
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120623T054003Z~65b2f2d2e7+unclean~20120623T083401Z
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 265
v=0
o=FreeSWITCH 1340907341 1340907343 IN IP4 224.168.168.168
s=FreeSWITCH
c=IN IP4 224.168.168.168
t=0 0
a=sendonly
m=audio 34567 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
我使用另一个软电话在该地址和端口上组播声音(通过wireshark验证).为什么我听不到声音?
顺便说一句,我使用xlite的软电话,服务器是freeswitch.
| 归档时间: |
|
| 查看次数: |
12108 次 |
| 最近记录: |