Thread: [OpenSIPStack] is it necessary to parse RTP in openSBC
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From: Ashish K. <ash...@gm...> - 2007-08-08 23:11:07
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Hi, I am a new user of openSBC product. I was just going through it and have a quiestion. Why it is necessary to parse RTP. Why cannot we use only the IP and port in the source and destination fiels to route the messages. why such type of details are needed to be known by ssrc=490290042 ntp=2007/8/8-2:22:0.995224 rtp=76800 psent=2 osent=320 ssrc=490290042 fraction=0 lost=0 last_seq=131074 jitter=0 lsr=419:09:26.072dlsr=0 |
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From: Joegen E. B. <joe...@gm...> - 2007-08-09 04:29:48
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There are cases where the SSRC may change in mid call. One example is when called is transferred and OpenSBC is configured for Local REFER. There are UAs that will ignore packets when SSRC is changed. Creating new packets gets rid of this problem. Ashish Khare wrote: > Hi, > > I am a new user of openSBC product. > I was just going through it and have a quiestion. > > Why it is necessary to parse RTP. Why cannot we use only the IP and port in > the source and destination fiels to route the messages. why such type of > details are needed to be known by > ssrc=490290042 ntp=2007/8/8-2:22:0.995224 rtp=76800 psent=2 osent=320 > ssrc=490290042 fraction=0 lost=0 last_seq=131074 jitter=0 > lsr=419:09:26.072dlsr=0 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
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From: Ashish K. <ash...@gm...> - 2007-08-09 16:58:54
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Hi Baclor, This is still not clear to me. Lets take a example: Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which handles only SIP signaling messages. Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and C are talking. But still they are abel to talk. Then how this case is different from yours. Can you please elaborate and explain to me. we are considering to build ALG. We have our own SIP stack ( Proxy and B2BUA ) but we dont have RTP stack. Also we dont want to parse the RTP stream. Just Rely it, based on source and destination IP and ports. Is this feasible ?. We are also exploring your openSBC if we can used it. On 8/9/07, Joegen E. Baclor <joe...@gm...> wrote: > > There are cases where the SSRC may change in mid call. One example is > when called is transferred and OpenSBC is configured for Local REFER. > There are UAs that will ignore packets when SSRC is changed. Creating > new packets gets rid of this problem. > > > Ashish Khare wrote: > > Hi, > > > > I am a new user of openSBC product. > > I was just going through it and have a quiestion. > > > > Why it is necessary to parse RTP. Why cannot we use only the IP and > port in > > the source and destination fiels to route the messages. why such type of > > details are needed to be known by > > ssrc=490290042 ntp=2007/8/8-2:22:0.995224 rtp=76800 psent=2 osent=320 > > ssrc=490290042 fraction=0 lost=0 last_seq=131074 jitter=0 > > lsr=419:09:26.072dlsr=0 > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Splunk Inc. > > Still grepping through log files to find problems? Stop. > > Now Search log events and configuration files using AJAX and a browser. > > Download your FREE copy of Splunk now >> http://get.splunk.com/ > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Joegen E. B. <joe...@gm...> - 2007-08-10 02:26:53
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inline... Ashish Khare wrote: > Hi Baclor, > This is still not clear to me. > Lets take a example: > Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which > handles only SIP signaling messages. > Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and > C are talking. > But still they are abel to talk. > Then how this case is different from yours. Can you please elaborate > and explain to me. There are two ways OpenSBC handles REFER. The default is to relay the REFER to the UA and let the UA do the transfer request. This is ok because the UA knows that there will be a change in the audio session. The second way (Local REFER) will not relay the REFER. Instead OpenSBC do the transfer. This leaves the other UA to not know that the call is actually transfered. If the transfer succeeded, a new media would with a different SSRC would have been created. If OpenSBC just relays that, the UA may reject the packets because the ssrc has already changed. > > we are considering to build ALG. We have our own SIP stack ( Proxy and > B2BUA ) but we dont have RTP stack. Also we dont want to parse the > RTP stream. Just Rely it, based on source and destination IP and > ports. Is this feasible ?. We are also exploring your openSBC if we > can used it. Of course this is feasible. You will have to change some lines of code in the media interface but it wont take much. Just post questions about the code if you need to clarify something. > > > > |
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From: Whit T. <wh...@wh...> - 2007-08-10 03:13:42
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Hey Guys, I was going to post this on the ATLSIP list, but it bounced back to me. (Maybe not working yet?) Anyway, I've been using ATLSIP for some time now, with pretty good success. I'm currently having a very bizarre issue. I've built a C# softphone and it runs fine on XP Pro machines. However, when it's installed on XP Home boxes, it sporadically crashes, giving the windows Send/Don't Send big report. The module which is affected and causing the crash is "ntdll.dll" . The softphone typically works for 2-3 calls, then crashes. Has anyone experienced this? I've compiled ATLSIP from CVS yesterday. Any suggestions on how to debug or even a possible solution for this would be great! Whit |
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From: Andre S. <eds...@ya...> - 2007-08-10 04:45:24
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Build your softphone on XP Home and make an installer for it. Whit Thiele <wh...@wh...> wrote: Hey Guys, I was going to post this on the ATLSIP list, but it bounced back to me. (Maybe not working yet?) Anyway, I've been using ATLSIP for some time now, with pretty good success. I'm currently having a very bizarre issue. I've built a C# softphone and it runs fine on XP Pro machines. However, when it's installed on XP Home boxes, it sporadically crashes, giving the windows Send/Don't Send big report. The module which is affected and causing the crash is "ntdll.dll" . The softphone typically works for 2-3 calls, then crashes. Has anyone experienced this? I've compiled ATLSIP from CVS yesterday. Any suggestions on how to debug or even a possible solution for this would be great! Whit ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. |
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From: Ilian J. C. P. <ip...@so...> - 2007-08-10 07:19:38
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Hi Whit, I'm on a XP Pro box today so I can't reproduce this yet. How are you deploying your softphone to XP Home? If it's via a VS Setup project, try packaging it with the appropriate CRT and ATL merge modules (and the policies that come with them). Regards, Ilian Whit Thiele wrote: > Hey Guys, > > I was going to post this on the ATLSIP list, but it bounced back to me. > (Maybe not working yet?) > > > > Anyway, I've been using ATLSIP for some time now, with pretty good success. > I'm currently having a very bizarre issue. I've built a C# softphone and it > runs fine on XP Pro machines. However, when it's installed on XP Home boxes, > it sporadically crashes, giving the windows Send/Don't Send big report. > > The module which is affected and causing the crash is "ntdll.dll" . The > softphone typically works for 2-3 calls, then crashes. Has anyone > experienced this? I've compiled ATLSIP from CVS yesterday. Any suggestions > on how to debug or even a possible solution for this would be great! > > > Whit > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Whit T. <de...@wh...> - 2007-08-10 13:45:32
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Ilian, I created a VS deployment project very similar to the OSS phone. I've also included most of the CRT/ATL modules I think I need. I'm assuming that since it works for a couple calls that side of things should be fine. Is there a complete list of the modules which should be included? Is there a DLL conflict with certain systems? I've used OSSPhone as a reference project for deployment. I wouldn't have thought there would be a huge difference between XP Pro and XP Home. I've been running this softphone on my dev box, but of course it may have many more of the libraries already installed because of the development enviroments. So far I've tried: Completely updated the machine with Windows Update (in case) Repaired .NET 2.0 Wrapping some exception handling code in the softphone But no luck so far. The only other thing I think may be causing this is not wrapping a Mutex around my MakeCall function like OSS phone does. Perhaps too many events are being fired at the same time. (I'm totally guessing) Let me know if you have suggestions or find anything out! Whit -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Ilian Jeri C. Pinzon Sent: Friday, August 10, 2007 2:20 AM To: ope...@li... Subject: Re: [OpenSIPStack] ATLSIP and ntdll.dll crashing Hi Whit, I'm on a XP Pro box today so I can't reproduce this yet. How are you deploying your softphone to XP Home? If it's via a VS Setup project, try packaging it with the appropriate CRT and ATL merge modules (and the policies that come with them). Regards, Ilian Whit Thiele wrote: > Hey Guys, > > I was going to post this on the ATLSIP list, but it bounced back to me. > (Maybe not working yet?) > > > > Anyway, I've been using ATLSIP for some time now, with pretty good success. > I'm currently having a very bizarre issue. I've built a C# softphone and it > runs fine on XP Pro machines. However, when it's installed on XP Home boxes, > it sporadically crashes, giving the windows Send/Don't Send big report. > > The module which is affected and causing the crash is "ntdll.dll" . The > softphone typically works for 2-3 calls, then crashes. Has anyone > experienced this? I've compiled ATLSIP from CVS yesterday. Any suggestions > on how to debug or even a possible solution for this would be great! > > > Whit > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: Ilian J. C. P. <ip...@so...> - 2007-08-13 02:38:55
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Whit, Can I have a copy of your log (sip.log) after ATLSIP crashes? - Ilian Whit Thiele wrote: > Ilian, > > I created a VS deployment project very similar to the OSS phone. I've also > included most of the CRT/ATL modules I think I need. I'm assuming that since > it works for a couple calls that side of things should be fine. Is there a > complete list of the modules which should be included? Is there a DLL > conflict with certain systems? I've used OSSPhone as a reference project for > deployment. I wouldn't have thought there would be a huge difference between > XP Pro and XP Home. I've been running this softphone on my dev box, but of > course it may have many more of the libraries already installed because of > the development enviroments. > > So far I've tried: > > Completely updated the machine with Windows Update (in case) > Repaired .NET 2.0 > Wrapping some exception handling code in the softphone > > But no luck so far. > > The only other thing I think may be causing this is not wrapping a Mutex > around my MakeCall function like OSS phone does. Perhaps too many events are > being fired at the same time. (I'm totally guessing) > > Let me know if you have suggestions or find anything out! > > > Whit > > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of Ilian > Jeri C. Pinzon > Sent: Friday, August 10, 2007 2:20 AM > To: ope...@li... > Subject: Re: [OpenSIPStack] ATLSIP and ntdll.dll crashing > > Hi Whit, > > I'm on a XP Pro box today so I can't reproduce this yet. How are you > deploying your softphone to XP Home? If it's via a VS Setup project, try > packaging it with the appropriate CRT and ATL merge modules (and the > policies that come with them). > > Regards, > Ilian > > Whit Thiele wrote: > >> Hey Guys, >> >> I was going to post this on the ATLSIP list, but it bounced back to me. >> (Maybe not working yet?) >> >> >> >> Anyway, I've been using ATLSIP for some time now, with pretty good >> > success. > >> I'm currently having a very bizarre issue. I've built a C# softphone and >> > it > >> runs fine on XP Pro machines. However, when it's installed on XP Home >> > boxes, > >> it sporadically crashes, giving the windows Send/Don't Send big report. >> >> The module which is affected and causing the crash is "ntdll.dll" . The >> softphone typically works for 2-3 calls, then crashes. Has anyone >> experienced this? I've compiled ATLSIP from CVS yesterday. Any suggestions >> on how to debug or even a possible solution for this would be great! >> >> >> Whit >> >> >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Ashish K. <ash...@gm...> - 2007-08-10 05:34:35
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Hi Baclor, Thanks for the reply. Is there any design document about openSBC, which will tell me in detail about how it is implementing the NAtting/ ALG functinality and how it will handle the Media streams. For Call Transfer, we will use the relay approach. Also, if i want to just relay the Media packets, can you let me know the algorithm you have applied in the openSBC. Also, in openSBC product, is High Availability supported or it is in roadmap ? On 8/10/07, Joegen E. Baclor <joe...@gm...> wrote: > > inline... > > > Ashish Khare wrote: > > Hi Baclor, > > This is still not clear to me. > > Lets take a example: > > Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which > > handles only SIP signaling messages. > > Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and > > C are talking. > > But still they are abel to talk. > > Then how this case is different from yours. Can you please elaborate > > and explain to me. > > There are two ways OpenSBC handles REFER. The default is to relay the > REFER to the UA and let the UA do the transfer request. This is ok > because the UA knows that there will be a change in the audio session. > The second way (Local REFER) will not relay the REFER. Instead OpenSBC > do the transfer. This leaves the other UA to not know that the call is > actually transfered. If the transfer succeeded, a new media would with > a different SSRC would have been created. If OpenSBC just relays that, > the UA may reject the packets because the ssrc has already changed. > > > > > > we are considering to build ALG. We have our own SIP stack ( Proxy and > > B2BUA ) but we dont have RTP stack. Also we dont want to parse the > > RTP stream. Just Rely it, based on source and destination IP and > > ports. Is this feasible ?. We are also exploring your openSBC if we > > can used it. > > > > Of course this is feasible. You will have to change some lines of code > in the media interface but it wont take much. Just post questions > about the code if you need to clarify something. > > > > > > > > > > > |
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From: Ashish K. <ash...@gm...> - 2007-08-14 19:46:46
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Hi Joegen, Please reply for my below mail. Thanks for the reply. Is there any design document about openSBC, which will tell me in detail about how it is implementing the NAtting/ ALG functinality and how it will handle the Media streams. For Call Transfer, we will use the relay approach. Also, if i want to just relay the Media packets, can you let me know the algorithm you have applied in the openSBC. Also, in openSBC product, is High Availability supported or it is in roadmap ? On 8/10/07, Ashish Khare <ash...@gm...> wrote: > > Hi Baclor, > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in detail > about how it is implementing the NAtting/ ALG functinality and how it will > handle the Media streams. > For Call Transfer, we will use the relay approach. > Also, if i want to just relay the Media packets, can you let me know the > algorithm you have applied in the openSBC. > > Also, in openSBC product, is High Availability supported or it is in > roadmap ? > > > On 8/10/07, Joegen E. Baclor <joe...@gm...> wrote: > > > > inline... > > > > > > Ashish Khare wrote: > > > Hi Baclor, > > > This is still not clear to me. > > > Lets take a example: > > > Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which > > > handles only SIP signaling messages. > > > Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and > > > C are talking. > > > But still they are abel to talk. > > > Then how this case is different from yours. Can you please elaborate > > > and explain to me. > > > > There are two ways OpenSBC handles REFER. The default is to relay the > > REFER to the UA and let the UA do the transfer request. This is ok > > because the UA knows that there will be a change in the audio session. > > The second way (Local REFER) will not relay the REFER. Instead OpenSBC > > do the transfer. This leaves the other UA to not know that the call is > > actually transfered. If the transfer succeeded, a new media would with > > > > a different SSRC would have been created. If OpenSBC just relays that, > > the UA may reject the packets because the ssrc has already changed. > > > > > > > > > > we are considering to build ALG. We have our own SIP stack ( Proxy and > > > > > B2BUA ) but we dont have RTP stack. Also we dont want to parse the > > > RTP stream. Just Rely it, based on source and destination IP and > > > ports. Is this feasible ?. We are also exploring your openSBC if we > > > can used it. > > > > > > > > Of course this is feasible. You will have to change some lines of code > > in the media interface but it wont take much. Just post questions > > about the code if you need to clarify something. > > > > > > > > > > > > > > > > > > > |
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From: Joegen E. B. <joe...@gm...> - 2007-08-15 03:13:01
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Ashish Khare wrote: > Hi Joegen, > Please reply for my below mail. > > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in detail > about how it is implementing the NAtting/ ALG functinality and how it will > handle the Media streams. No there isn't yet. Your best bet is to analyze the code and ask questions. > For Call Transfer, we will use the relay approach. Then you may not need to parse RTP > Also, if i want to just relay the Media packets, can you let me know the > algorithm you have applied in the openSBC. > RTP is parsed in RTPSession::OnReceiveData(). You may override this function to not parse anything and just return e_ProcessPacket for all cases. I haven't done this myself so i'm not sure what side effects there will be. If that happens, you are on your own. > Also, in openSBC product, is High Availability supported or it is in > roadmap > ? > It is planned but did not get a date yet in the roadmap. > > On 8/10/07, *Ashish Khare* <ash...@gm... > <mailto:ash...@gm...>> wrote: > > Hi Baclor, > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in > detail about how it is implementing the NAtting/ ALG functinality > and how it will handle the Media streams. > For Call Transfer, we will use the relay approach. > Also, if i want to just relay the Media packets, can you let me > know the algorithm you have applied in the openSBC. > > Also, in openSBC product, is High Availability supported or it is > in roadmap ? > > > On 8/10/07, *Joegen E. Baclor* <joe...@gm... > <mailto:joe...@gm...>> wrote: > > inline... > > > Ashish Khare wrote: > > Hi Baclor, > > This is still not clear to me. > > Lets take a example: > > Sip Client A is talking to Sip Client B through > Proxy/B2BUA(P) which > > handles only SIP signaling messages. > > Now in Call, Sip Client A is trasnferred to Sip CLient C and > now B and > > C are talking. > > But still they are abel to talk. > > Then how this case is different from yours. Can you please > elaborate > > and explain to me. > > There are two ways OpenSBC handles REFER. The default is to > relay the > REFER to the UA and let the UA do the transfer request. This > is ok > because the UA knows that there will be a change in the audio > session. > The second way (Local REFER) will not relay the REFER. > Instead OpenSBC > do the transfer. This leaves the other UA to not know that > the call is > actually transfered. If the transfer succeeded, a new media > would with > a different SSRC would have been created. If OpenSBC just > relays that, > the UA may reject the packets because the ssrc has already > changed. > > > > > > we are considering to build ALG. We have our own SIP stack ( > Proxy and > > B2BUA ) but we dont have RTP stack. Also we dont want to > parse the > > RTP stream. Just Rely it, based on source and destination IP and > > ports. Is this feasible ?. We are also exploring your openSBC > if we > > can used it. > > > > Of course this is feasible. You will have to change some > lines of code > in the media interface but it wont take much. Just post > questions > about the code if you need to clarify something. > > > > > > > > > > > > |
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