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Monday, November 17, 2008

Updated Sip stack and UA for Android SDK 1.0r1

Folks, my company just released an updated version of the SIP stack and UA (MJSIP) ported for the latest SDK 1.0r1. This is the same SDK that G1 uses. This is an update to this post.

The new SDK changes several things in the UI and well as some APIs. Kudos to the team lead Nitin Khanna (first.last at hsc.com) for continuing the commitment to android. 

Grab a copy of the latest version from here.

25 comments:

  1. java.io.FileNotFoundException: /data/data/org.hsc.sip.ua/files/127.0.0.1.5060_messages.log

    correlates with "Error reading configuration file" dialog.

    Looking into this.

    W

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  2. Hi!

    Please follow the code install instructions as suggested in the accompanying document. You have to create the directory

    /data/data/org.hsc.sip.ua/files


    as described in the installation procedures.

    Regards
    Nitin

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  3. [...] Nov-20-2008: Updated SDK + UA for 1.0 SDK released here. Also released jSTUN port for android [...]

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  4. What installation procedures?
    How do you create this directory?
    When I try to mkdir I get access denied, and when I use DDMS I can't see inside the /data dir.

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  5. FYI, this release of the SIP UA code says:
    "This release does not include the RTP stack as it is already a part of the released Android source code..."

    There does not appear to be an RTP stack built in to Android; someone please correct me if I am wrong.

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  6. There is an RTP stack on Android, but it's not currently exposed through the Java API. I'm porting JNA to Android to try to get simplified native access to the RTP stack (among other things).

    Sadly, depending on JNA means getting a non-java library onto the phone, so if the softphone I'm working on requires JNA, it won't be an easy-to-install program. You could only install it on an unlocked phone or on a developer phone.

    One step at a time...

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  7. I have managed to compile and run this on a G1 device, I had initial problems being able to login to any SIP server, I overcame this by installing a local SIP server for testing. Other sip clients can login and make voice calls, using the SIP client in the build I am able to connect, receive calls if it is set to auto-answer and make outgoing calls to other sip clients and PSTN devices, at this stage there is no audio being sent or being received during these calls.

    Is there any plans to continue with the development of the sip client?

    Regards

    Mark

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  8. Hi PV,

    The installation procedures are same as the ones specified for the initial release of Sipdroid. All you have to do is to unpack the the archive and import the source files or open the project using eclipse. Build and you are good to go.

    Please let me know in case you are still facing issues relating to the same.

    Regards
    Nitin

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  9. Hi Mark,

    Thanks for the update. Yes, Sipdroid works on actual device with some minor tweaks. I had been busy lately and have not been able to update the release in the full swing. However, the changes are minor and are more related to changing the file path.

    We are currently working on extending and demonstrating an IMS client using SipDroid. You should hear from us soon.

    Regards
    Nitin

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  10. Hi Ross,

    We are uploading the ported version of a java based RTP stack on our website shortly. This will be not as efficient as a native stack which is present in the Android Source code, however, we believe that it still should be a good library to start building some prototypes using Sipdroid and Media.

    Regards
    Nitin

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  11. Hi Nitin,

    You said in your comments before that 'We are currently working on extending and demonstrating an IMS client using SipDroid. You should hear from us soon', do you have any time lines committed towards this? When can we expect this?

    Using this IMS client can we interact with Yahoo! Messenger?

    Thanks
    Malli

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  12. Hi Nitin,
    You mentioned that you have a Java RTP stack for Android, Can you please send me the link?

    Cheers,
    Earlence

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  13. @Malli,
    Sorry, no timeframes yet. The solution will be tested against open IMS. AFAIK, Y! does not use IMS extensions for SIP.

    ReplyDelete
  14. Hi:
    I have downloaded the Sipdroid and run it by emulator, but I have a problem now. The problem is I cannot registering to SIP server. I used Wireshark to view packets. I have seen the first request packet from emulator to sip server and the response packet from sip server to emulator. After that, the emulator didn't send the second register (with authentication) packet to sip server, so did the 200 OK packet from sip server to emulator. Can you give me some help?

    Thanks in advance.

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  15. I know I'm whining, but do you have any idea when this IMS client could be released?

    Thanks,
    A fan

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  16. Hi Nitin,

    Is the IMS client still being worked on?

    ReplyDelete
  17. Yes, its actually ready. I'll post it next week if I am back from my travel.

    ReplyDelete
  18. Okay folks, you can download the IMS UA from HERE

    ReplyDelete
  19. Hi Susan!

    Sorry for being delayed on this. Using the ADB SHELL tool simply create a directory called files in data/data/org.hsc.ua.sip/ folder and things should start working for you.

    This has also been captured on a number of previous posts.

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  20. Hi nitin,
    Can you tell me where to get the source code of sipdroid?
    I've found it nowhere.
    Thanks.

    ReplyDelete
  21. Sorry, my mistake, I got the source code.
    I thought it's here.

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  22. Has anyone taken this and tried to port it to the Blackberry?

    ReplyDelete
  23. hi Arjun!

    i have a question: why isn't the parseLine method in the Configure class implemented? How do the settings get read from the configuration file?
    Am i missing something here?

    ty!
    asy

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  24. Hi Arjun,

    I am not able to register SIPROID_UA with Brekeke ONDO SIP Server.
    I added all the configuration settings. But was not able to do the same.
    Also i am trying to write a mini application on android to register with the SIP server. But i am not able to do that.
    Here is the request that goes
    REGISTER sip:172.21.38.33 SIP/2.0
    Call-ID: d831ee9997aa8e86eb6ca1408ae4053e@127.0.0.1
    CSeq: 1 REGISTER
    From: ;tag=12345
    To:
    Via: SIP/2.0/UDP 127.0.0.1:9005;branch=z9hG4bK04c2fa65c0bf7073adfbd2d5cb455142
    Max-Forwards: 70
    Contact: ;expires=3600
    Route:
    User-Agent: Jaon SIP POC1.0CVS-Wed_Jul_15_11-20-20_GMT+05-30_2009
    Content-Length: 0

    But i dont receive any response and it simpy timeout. Can you suggest me something for this issue/ I am using Jain SIP.

    ReplyDelete