Hi, I have created two different accounts. When I make a loop call (to local uri where pjsip is running) for test purposes with one of the created accounts, the default account is matched with the call. Is this specific to loop calls or am I making a mistake? account 1 status = pjsua_acc_add_local(udp_transport_id,0,&app_acc1_id); account 2 status = pjsua_acc_add_local(udp_transport_id,0,&app_acc2_id); I specify the account when making a call, but only the default account accepts calls. Thank you for your help. --- On Wed, 5/12/10, pjsip-request at lists.pjsip.org <pjsip-request at lists.pjsip.org> wrote: From: pjsip-request@xxxxxxxxxxxxxxx <pjsip-request at lists.pjsip.org> Subject: pjsip Digest, Vol 33, Issue 13 To: pjsip at lists.pjsip.org Received: Wednesday, May 12, 2010, 5:48 AM Send pjsip mailing list submissions to ??? pjsip at lists.pjsip.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org or, via email, send a message with subject or body 'help' to ??? pjsip-request at lists.pjsip.org You can reach the person managing the list at ??? pjsip-owner at lists.pjsip.org When replying, please edit your Subject line so it is more specific than "Re: Contents of pjsip digest..." Today's Topics: ???1. RTP over TCP in pjsip (abbas abdolali pour) ???2. Re: Turn your free SIP softphone into a voice quality ? ? ? monitoring instrument with Sevana?s NIQA application (Klaus Darilion) ???3. Re: Big Problem SIP traffic Blocking.. Is thre any great ? ? ? solution (amr at ntstel.com) ???4. Re: Voice Distorted on 2G iPhones (Re Mo) ???5. Re: Turn your free SIP softphone into a voice quality ? ? ? monitoring instrument with Sevana?s NIQA application (Sevana Oy) ???6. Re: Voice Distorted on 2G iPhones (Mukesh Sharma) ---------------------------------------------------------------------- Message: 1 Date: Tue, 11 May 2010 23:34:08 -0700 (PDT) From: abbas abdolali pour <aaa_pour@xxxxxxxxx> To: pjsip at lists.pjsip.org Subject: RTP over TCP in pjsip Message-ID: <641528.23846.qm at web51601.mail.re2.yahoo.com> Content-Type: text/plain; charset=us-ascii HI all I check pjmedia and transport_udp.c for TCP connection. RTP over TCP implantation are exist or not? if no What kind of consideration should be added to transport layer to implementation the RTP over TCP . Best Abbas ? ? ? ------------------------------ Message: 2 Date: Wed, 12 May 2010 09:50:04 +0200 From: Klaus Darilion <klaus.mailinglists@xxxxxxxxx> To: pjsip list <pjsip at lists.pjsip.org> Subject: Re: Turn your free SIP softphone into a voice quality ??? monitoring instrument with Sevana?s NIQA application Message-ID: <4BEA5DAC.2010303 at pernau.at> Content-Type: text/plain; charset=windows-1252; format=flowed Hi! Am 12.05.2010 06:37, schrieb Sevana Oy: >> Does somebody know how pjsip writes the wavefile? Will it be written >> exactly like to the audio device (with possible jitter buffer >> under/overrun and playback-speed adjustments) or will the voice sample >> be written just one after the other to the wave file? > > Call audio will be saved one after another into the same file, however, > this can also be solved in order to receive recording of a single call. That's not what I asked - maybe I should make myself more clear: During a normal phone call, the receiver may do manipulations to the audio stream before playing back the audio to the user. For example SIP clients often have dynamical jitter buffer - when the buffer gets empty the playback speed will be reduced, when the buffer gets full the playback speed will be increased, old packets may be ignored completely. When a call is recorded, this manipulations are not needed because it doesn't matter if packets arrive late as for recording there are no real-time constraints. The receiver can wait until the RTP packets are received and then it saves the audio payload without need for manipulation to the wav file. If a client writes a wav file like described above, and the is no packet loss - the recorded file will always be identical to the file sent by the other party. If a client does audio manipulation also for recordings, then the recorded file will differ from the original and MOS should be different. regards Klaus ------------------------------ Message: 3 Date: Wed, 12 May 2010 03:32:59 -0500 (CDT) From: amr@xxxxxxxxxx To: "pjsip list" <pjsip at lists.pjsip.org> Subject: Re: Big Problem SIP traffic Blocking.. Is thre any ??? great solution Message-ID: ??? <1924.117.58.242.30.1273653179.squirrel at webmail.ntstel.com> Content-Type: text/plain; charset="iso-8859-1" Hi, Do you have a how to for configuring IPsec server in a windows 2003 server machine ? Riad. > The voice performance would be terrible over HTTP. > > Rather than hacking up the sip and RTP stacks with a custom solution, I'd > look at using solutions that are already out there.? SIPTLS on a > non-standard port and IPsec for the RTP traffic, or IPsec for both > signaling and media.? You can't inspect the traffic if it is encrypted and > IPsec can be used to obfuscate call traffic heuristics. > > > On May 10, 2010, at 11:30 PM, Varun Singh wrote: > >> HI All. >> >> These we have a big problem in the gulf for the telecom companies are >> facing that is they are incapable of routing the sip traffic through the >> internet. AFAIK the sip traffic is blocked not only by the standard port >> blocking for SIP and RTP but also Deep Packet Analyzers are being used. >> The DPA(Deep Packet Analyzers) algorithms are read the packets and can >> also determine the nature of communication going on. That is why in some >> areas the OpenVPN Tunneling on PJSIP is also being caught and blocked. >> >> Other solution can be HTTP tunneling which can be great solution but I >> don't think it will be a performance oriented as being a TCP oriented >> protocol. But it is the most effective solution. So is there any >> implementation available for the HTTP Tunneling with the PJSIP so that >> can be implemented. Is there are other solutions for it. >> >> >> Please propose so that we can start finding the best solution. >> >> >> Regards: >> varun >> >> DISCLAIMER: >> ----------------------------------------------------------------------------------------------------------------------- >> >> The contents of this e-mail and any attachment(s) are confidential and >> intended for the named recipient(s) only. >> It shall not attach any liability on the originator or HCL or its >> affiliates. Any views or opinions presented in >> this email are solely those of the author and may not necessarily >> reflect the opinions of HCL or its affiliates. >> Any form of reproduction, dissemination, copying, disclosure, >> modification, distribution and / or publication of >> this message without the prior written consent of the author of this >> e-mail is strictly prohibited. If you have >> received this email in error please delete it and notify the sender >> immediately. Before opening any mail and >> attachments please check them for viruses and defect. >> >> ----------------------------------------------------------------------------------------------------------------------- >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100512/d89aab6a/attachment-0001.html> ------------------------------ Message: 4 Date: Wed, 12 May 2010 14:59:40 +0300 From: Re Mo <remo9071@xxxxxxxxx> To: Jeff Brower <jbrower at signalogic.com> Cc: pjsip at lists.pjsip.org Subject: Re: Voice Distorted on 2G iPhones Message-ID: ??? <AANLkTimfjxiRssiY95yOGJwAKRBbmu6Apoety_jABVrR at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Dear Jeff 1) The example below indeed shows PCMU. I have used this for debugging purposes. Sorry for not mentioning this. 2) Regarding the bandwidth - since the app calls over WiFi and not data, I think this should not be an issue. What do you think? Thx Remo On Wed, May 12, 2010 at 2:24 AM, Jeff Brower <jbrower at signalogic.com> wrote: > Remo- > > > The bad call quality on the 2G iPhone is a combination of static noise > (not > > too loud), 'robotic' speech and some short? missing bits. > > > > I gathered some info provided by PJSIP, regarding a single call I made in > > case it's helpful: > > > > Call time: 00h:01m:16s > > 1st res in 3667 ms > > conn in 14909ms > > SRTP status: Not active > > Crypto-suite: (null)\n #0 PCMU @8KHz > > , sendrecv, > > RX pt=0 > > > > stat last update: 00h:00m:01.497s ago > > total 4.3Kpkt 698.8KB (873.6KB IP hdr) @avg=63.4Kbps/79.3Kbps > > *pkt loss=26 (0.6%)* > > discrd=0 (0.0%) > > dup=0 (0.0%) > > reord=0 (0.0%) (msec) min avg max last dev > > *loss period: 20.000 20.000 20.000 20.000 0.000* > > jitter : 0.000 4.626 194.625 24.125 5.505 > > TX pt=0 > > ptime=20ms > > One thing that seems to jump out is the call is not actually using G729, as > you had mentioned.? Instead it's using > G711 uLaw.? The aggregate bitrate (including IP/UDP/RTP headers) is almost > 80 kbps, which would be correct for 20 msec > G711.? However, 80 kbps is too high for your 2G network max bandwidth, at > least according to the bandwidth figures you > gave previously. > > -Jeff > > > On Mon, May 10, 2010 at 4:09 PM, Jeff Brower <jbrower at signalogic.com> > wrote: > > > >> Remo- > >> > >> > My iPhone app uses the PJSIP iPhone Audio driver. > >> > It works well on 3G & 3GS iPhones, but I am having some voice quality > >> issues > >> > on 2G iPhones. > >> > When making a call with a 2G iPhone, the voice is distorted and it is > >> > impossible to have a conversation. > >> > > >> > Did anyone experince the same issue or has any advise? this can be > >> related > >> > to PJSIP/PJSIP audio driver or the 2G iPHone CPU. > >> > >> 2nd-gen iPhones don't have as much CPU horsepower, but that's probably > not > >> the whole issue.? What does the distortion > >> sound like?? Missing packets?? Static?? "Cyborg voice"? > >> > >> -Jeff > >> > >> > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100512/37cafee6/attachment-0001.html> ------------------------------ Message: 5 Date: Wed, 12 May 2010 16:36:53 +0400 From: "Sevana Oy" <sales@xxxxxxxxx> To: "Klaus Darilion" <klaus.mailinglists at pernau.at>,??? "pjsip list" ??? <pjsip at lists.pjsip.org> Subject: Re: Turn your free SIP softphone into a voice quality ??? monitoring instrument with Sevana?s NIQA application Message-ID: <A194167A20CC4B75BF34C4C2A18BF6DC at laptop> Content-Type: text/plain; format=flowed; charset="windows-1252"; ??? reply-type=response Hi Klaus, If I get your question right this time the point is in the perceptual model utilized in a voice quality assessment software. Basically, the best way is to use both: MOS generated by our software or P.862/P.563 together with typical VoIP characteristics. Our software like ITU standards (if I may compare them here) implement a perceptual voice quality assessment model that produces MOS scores according to how a human percepts the audio thus if packets were lost or "recovered" that will lead to MOS score decrease or increase. Hope I answered to your comment. Thanks! ----- Original Message ----- From: "Klaus Darilion" <klaus.mailinglists@xxxxxxxxx> To: "pjsip list" <pjsip at lists.pjsip.org> Cc: "Sevana Oy" <sales at sevana.fi> Sent: Wednesday, May 12, 2010 11:50 AM Subject: Re: Turn your free SIP softphone into a voice quality monitoring instrument with Sevana?s NIQA application > Hi! > > Am 12.05.2010 06:37, schrieb Sevana Oy: >>> Does somebody know how pjsip writes the wavefile? Will it be written >>> exactly like to the audio device (with possible jitter buffer >>> under/overrun and playback-speed adjustments) or will the voice sample >>> be written just one after the other to the wave file? >> >> Call audio will be saved one after another into the same file, however, >> this can also be solved in order to receive recording of a single call. > > That's not what I asked - maybe I should make myself more clear: > > During a normal phone call, the receiver may do manipulations to the audio > stream before playing back the audio to the user. For example SIP clients > often have dynamical jitter buffer - when the buffer gets empty the > playback speed will be reduced, when the buffer gets full the playback > speed will be increased, old packets may be ignored completely. > > When a call is recorded, this manipulations are not needed because it > doesn't matter if packets arrive late as for recording there are no > real-time constraints. The receiver can wait until the RTP packets are > received and then it saves the audio payload without need for manipulation > to the wav file. > > If a client writes a wav file like described above, and the is no packet > loss - the recorded file will always be identical to the file sent by the > other party. > > If a client does audio manipulation also for recordings, then the recorded > file will differ from the original and MOS should be different. > > regards > Klaus > ------------------------------ Message: 6 Date: Wed, 12 May 2010 18:20:30 +0530 From: Mukesh Sharma <mukesh.s@xxxxxxxxxxxx> To: pjsip list <pjsip at lists.pjsip.org> Subject: Re: Voice Distorted on 2G iPhones Message-ID: <E25F63CB-716B-484D-95D1-C53B5EAF992A at geodesic.com> Content-Type: text/plain; charset="us-ascii" hi, we have sorted out 2g Distorted voice issue. u can checkout code on spokn on git hub. pjsip compilation and changes is in? this folder lib/pjsipcompilation. u can check spokn which? is available on app store. Regards Mukesh On 12-May-2010, at 5:29 PM, Re Mo wrote: > Dear Jeff > > 1) The example below indeed shows PCMU. I have used this for debugging purposes. Sorry for not mentioning this. > 2) Regarding the bandwidth - since the app calls over WiFi and not data, I think this should not be an issue. What do you think? > > Thx > Remo > > On Wed, May 12, 2010 at 2:24 AM, Jeff Brower <jbrower at signalogic.com> wrote: > Remo- > > > The bad call quality on the 2G iPhone is a combination of static noise (not > > too loud), 'robotic' speech and some short? missing bits. > > > > I gathered some info provided by PJSIP, regarding a single call I made in > > case it's helpful: > > > > Call time: 00h:01m:16s > > 1st res in 3667 ms > > conn in 14909ms > > SRTP status: Not active > > Crypto-suite: (null)\n #0 PCMU @8KHz > > , sendrecv, > > RX pt=0 > > > > stat last update: 00h:00m:01.497s ago > > total 4.3Kpkt 698.8KB (873.6KB IP hdr) @avg=63.4Kbps/79.3Kbps > > *pkt loss=26 (0.6%)* > > discrd=0 (0.0%) > > dup=0 (0.0%) > > reord=0 (0.0%) (msec) min avg max last dev > > *loss period: 20.000 20.000 20.000 20.000 0.000* > > jitter : 0.000 4.626 194.625 24.125 5.505 > > TX pt=0 > > ptime=20ms > > One thing that seems to jump out is the call is not actually using G729, as you had mentioned.? Instead it's using > G711 uLaw.? The aggregate bitrate (including IP/UDP/RTP headers) is almost 80 kbps, which would be correct for 20 msec > G711.? However, 80 kbps is too high for your 2G network max bandwidth, at least according to the bandwidth figures you > gave previously. > > -Jeff > > > On Mon, May 10, 2010 at 4:09 PM, Jeff Brower <jbrower at signalogic.com> wrote: > > > >> Remo- > >> > >> > My iPhone app uses the PJSIP iPhone Audio driver. > >> > It works well on 3G & 3GS iPhones, but I am having some voice quality > >> issues > >> > on 2G iPhones. > >> > When making a call with a 2G iPhone, the voice is distorted and it is > >> > impossible to have a conversation. > >> > > >> > Did anyone experince the same issue or has any advise? this can be > >> related > >> > to PJSIP/PJSIP audio driver or the 2G iPHone CPU. > >> > >> 2nd-gen iPhones don't have as much CPU horsepower, but that's probably not > >> the whole issue.? What does the distortion > >> sound like?? Missing packets?? Static?? "Cyborg voice"? > >> > >> -Jeff > >> > >> > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > Regards, > > Mukesh Sharma > Head - Principal Technologist? | Spokn > Geodesic Limited | www.geodesic.com > Tel: +91 22? 40315859 > > Mobile: +919892029162 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100512/fb282a95/attachment.html> ------------------------------ _______________________________________________ pjsip mailing list pjsip at lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org End of pjsip Digest, Vol 33, Issue 13 ************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100512/f4e63579/attachment-0001.html>
