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SIP woes

Hi all,

I am experimenting with the Spark 2.5.2 and Openfire with Enterprise plugins for few days. The jabber part is great, as can be expected from the guys understand XMPP, but the SIP part is really bad!

  • First, why Spark have to get the SIP settings from the servers(through http://www.jivesoftware.com/protocol/sipark)? What’'s good with it? Why not let the user choose to create the new SIP account, just like the opportunity to create the Yahoo account? Just want to tie users to Enterprise plug-in for VoIP with Spark? Sorry there are tons of free softphones out there, even some support XMPP like www.openwengo.com or PhoneGaim.

  • And why you don’'t allow user to choose the sound devices and the codecs as well? If I have 2 mics, one with the webcam, and another one is standard mic, why I have to use the default? The same is for codecs. Why not let me choose?

  • Why Spark don’‘t allow for manual setting for the SIP and RTP port? It’'s can be a headache with some cheap ADSL router.

  • And the worst thing is the SIP implementation part. I got the ethereal log here but don’'t know the way to attach into this message. The symptom is as follow:

  • The spark register well with SIP server (in this case it’'s an Asterisk 1.2.1).

  • You can make a call with Spark.

  • When Spark hangup, none of BYE message was sent from Spark.

  • When other side hangup and Asterisk send BYE to Spark. Spark never ACK, so Asterisk have to send about 10 other BYEs and hangup by itself.

I don’‘t want to do further testings, there are too many issues. Just ask my self, why don’'t you at Jive give the SIP part the OS status?

With best regards,